KR100389453B1 - wireless-Internet based contents providing method for raising listening ability of foreign language - Google Patents

Abstract

Disclosed is a wireless Internet-based content service method for learning a foreign language. After accessing the site through the wireless Internet using a mobile communication terminal, the member authentication process is performed, and if the member is not a member, the member registration process is performed, and then the voice signal processing engine program is automatically installed and installed in the mobile communication terminal. do. Then, the list information of the foreign language listening ability learning content that can be provided is transmitted to the mobile communication terminal which has been certified by the member, and the content file selected by the member is transmitted to the mobile communication terminal for storage. When the user selects a specific content file from among foreign language listening ability learning contents stored in the mobile communication terminal and gives a playback command, the voice signal processing engine program specifies a playback time adjustment rate α for specifying the playback speed of audio data and a weak sound. It receives a differential amplification factor β for the input signal, receives a setting value, and processes and reproduces the audio data of the content file according to the set playback speed and the differential amplification factor.

Description

Wireless-Internet based contents providing method for raising listening ability of foreign language}

The present invention relates to a wireless Internet-based content learning method for foreign language learning, and more particularly, accesses a foreign language learning content providing site through a mobile communication terminal, downloads and stores desired content, and plays audio and / or text when necessary. The present invention relates to a method of improving the listening power of reproduction sound by performing a reproduction speed and a weak sound enhancement process during audio reproduction.

At present, the Internet is not limited to a specific field but is widely used, and as part of real life, the number of Internet users using the Internet is rapidly increasing. Accordingly, the number of service providers that provide specific services to Internet users is increasing rapidly. Examples of services based on the Internet include information providers that process and provide information such as securities, auctions, real estate, and commodity transactions, and chat rooms that provide communication points between Internet users. It is one of the popular representative services.

By the way, foreign language learning is a general form using a computer terminal connected to the wired Internet, which is a lot of time or space constraints. For busy workers, it's not hard to spend extra time at home after work to continue learning foreign languages. As a result, many office workers often use their commuting hours to learn foreign languages. Even in this movement, there is a demand for providing a service for facilitating foreign language learning, in particular, foreign language listening ability.

Of course, the foreign language listening ability learning has been made by a method of playing back a tape using a cassette tape recorder. However, this method has the inconvenience of carrying a cassette tape and a cassette tape recorder. In addition, the incidence of intellectual curiosity is weak because of the monotony of the content in that there is a limit on the number of tapes to be carried and passively listens to the contents recorded on the tape.

With the rapid spread of mobile communication services, attempts have been made to access the Internet using mobile communication terminals, one of which is the wireless Internet. The user of the mobile communication terminal can access the desired Internet site through the wireless Internet, investigate the necessary information, and can download the desired content. However, until now, the wireless Internet service is still in its infancy and only serves textual content, and in particular, there is no service for creating foreign language learning files that integrate audio data and text data and providing them as foreign language learning contents. None suggested.

On the other hand, the reproducing sound is often difficult to understand a foreign language, there may be a variety of causes for this. The reason may be that the original tone is weakly recorded or recorded at a very high speed, but it is also possible that the listener's hearing is weak and he / she does not hear it well, or that the hearing is fundamentally low due to the foreign language being played. This can be

Therefore, in order to effectively support learning of foreign language listening ability, simply reproducing audio data is not sufficient. It is necessary to provide a separate listening support function for learners with low listening ability.

As one solution, reinforcing the weak sound strength that is hard to hear may be one solution. In general, syllabic language such as English has a linguistic characteristic that can convey more information by accent and accent, unlike Korean or Japanese. Omitting or attenuating some of the various pronunciations in a word does not have much problem in conveying its meaning, so syllable languages, especially English pronunciations, are often treated as mute or weak. In contrast, in the case of nonsyllabic languages such as Korean or Japanese, communication through the voice is mainly responsible for the meaning included in each word, so the accent or accent does not take much weight. The pronunciation feature of non-syllable languages is that each word is pronounced correctly, and it is rarely pronounced as a silent or weak sound. It is very difficult for people who use non-syllable languages such as Korean or Japanese to listen to the accented syllable parts of English pronunciations, such as silent or weak sounds. As a result, those who are familiar with Korean or Japanese pronunciation systems needed to repeat listening lessons in order to understand English pronunciation correctly. It is very difficult to artificially add silence, so that it is possible to enhance the strength of the weak sound when the audio signal is reproduced. However, known gain control methods of audio signals are methods that apply the same amplification rate uniformly without depending on the intensity of sound, but rather are not actually employed because they degrade the quality of the reproduced sound. That is, according to these methods, if the amplification of the weak sound to the desired level by applying the same gain ratio uniformly without distinction between weak sound and strong sound, even the strong sound is strengthened, and the strong sound part is heard as a loud and torn sound. The quality of the reproduction sound is rather worse.

Alternatively, it may be necessary to make the playback speed slower if the original normal playback speed is felt relatively fast, or vice versa. Various techniques are known in the art for varying the reproduction speed. For example, since the overlap-addition (OLA) was introduced by Roucus and Wilgus in 1985, it changed to the synchronous overlap addition method (SOLA), the OLA method based on waveform similarity, that is, the WSOLA method. It is developing. In order to change the playback speed, the SOLA algorithm overlaps the audio data at regular intervals by using a window of a constant size, cuts out the corresponding block, and adds the audio signal by rearranging the blocks on the time axis in response to the required speed change. Processing. In order to prevent sound degradation, a technique of synthesizing two block signals by shifting the similarity of waveforms of overlapping portions by an interval corresponding to the largest value may be applied.

However, the technology of changing the reproduction speed of the audio signal can maintain good sound quality even after processing, and furthermore, it is highly desirable if it can be simultaneously processed in combination with a technique of varying the sound intensity. In addition, a weak sound reinforcement or variable control of playback speed may be provided to a mobile communication terminal and combined with a service for providing foreign language learning contents based on the wireless Internet will have a synergistic effect.

An object of the present invention is to provide a foreign language learning content when a subscriber having a mobile communication terminal capable of accessing the wireless Internet accesses the corresponding Internet site, and enhances the sound volume and adjusts the playing speed when playing the content file using the mobile communication terminal. By installing a program that can provide a function to the mobile communication terminal, the user can download the foreign language learning contents at any time desired by the user of the mobile communication terminal, or select the downloaded contents and play them according to their listening level. An object of the present invention is to provide a wireless Internet based content service method for learning a foreign language that can be effectively performed.

1 is a block diagram showing a schematic configuration of a system required to service foreign language learning content in a wireless Internet based on the present invention.

2 is a block diagram showing a schematic configuration of a mobile communication terminal.

3 is a software functional block diagram for realizing an audio signal processing method according to the present invention.

4 shows an appearance of a mobile communication terminal.

5 is a flowchart illustrating a procedure of downloading a foreign language learning content to a mobile communication terminal by accessing a content providing system using a wireless Internet.

6 is a flowchart illustrating a procedure of reproducing foreign language learning content stored in a mobile communication terminal.

7A to 7I illustrate screen configurations of a display panel of a mobile communication terminal when downloading and playing content for foreign language learning.

8 is a flowchart schematically illustrating a method of processing an audio signal applied when reproducing foreign language learning content in a mobile communication terminal.

FIG. 9 is a flowchart showing an algorithm for performing time-base conversion according to a reproduction time adjustment rate α set by a user in processing an audio signal.

FIG. 10 is a reference diagram for explaining the principle of obtaining the time-axis conversion signal y (·) by increasing or decreasing the length of the original audio signal x (·) corresponding to the magnitude of the reproduction time adjustment rate α set by the user. .

FIG. 11 is a flowchart illustrating an algorithm for performing differential amplification by applying an amplification factor higher than strong sound of an audio signal.

12 (a) to 12 (c) are graphs showing various examples of the differential amplification function applicable to the differential amplification of an audio signal.

13A to 13C show waveform diagrams of time of an original audio signal, a time-base converted audio signal, and a differentially amplified audio signal, respectively.

** Explanation of symbols for main parts of drawings **

2: mobile communication terminal 3: mobile communication service system

4: Wireless Internet Gateway Server 5: Web Server

6: Database Server 8: Content Provision System

12: playback time adjustment section 14: differential amplifier

16: filter section 18: buffer section

20: key input unit 22: voice signal processing unit

24: speaker 26: display panel driver

28: display panel 30: microprocessor unit (MPU)

32: radio transmission / reception processing unit (RF unit) 34: memory unit

36: ROM 38: Microphone

40: antenna

According to the present invention for achieving the above object, when accessing the site through the wireless Internet using a mobile communication terminal performs a member authentication procedure if the member is not a registered member and then the voice signal processing engine Transmitting a program and automatically installing the mobile communication terminal; And transmitting the list information of the foreign language listening ability learning contents that can be provided to a mobile communication terminal that has been authenticated by a member, and transmitting the selected content file to the mobile communication terminal to store the contents file. As a method for simultaneously controlling the reproduction speed of the audio data included in the content file according to the rate and the enhancement of the sound by the differential amplification,

The voice signal processing engine program may include a playback time adjustment rate α for designating a playback speed of the audio data when a user selects a specific content file from among foreign language listening ability learning contents stored in the mobile communication terminal and gives a playback command. A function for receiving a differential amplification ratio β for a strong sound of a weak sound and inputting a setting value thereof; The audio data x (·) is continuously read from the first memory in units of N samples, and the next frame F _m crosses a smaller number of samples than the N number at the start of the previous frame F _m-1 . A data reading function for reading a predetermined number of samples repeatedly between adjacent frames by reading from the sample of the jumped position; The next frame F _{m is added} to the second memory so as to overlap the previous frame F _m-1 already recorded as the time-base conversion signal y (·) by a predetermined number of samples, and the samples of the overlapped portion are weighted, added, and non-overlapping. The portion is copied as it is, and the number of samples of the overlapping portion is added in such a manner as to increase or decrease in correspondence with the size of the reproduction time adjustment rate α so as to obtain a signal y (·) whose time axis length is converted in proportion to the reproduction time adjustment rate α. Time axis conversion function; And amplifying by substituting the sample data x _i of each frame into a predetermined differential amplification function f (x _i ) corresponding to the differential amplification ratio β, in particular, the differential amplification function f (x _i ) gradually increases from weak to strong. Since the amplification gain is a function applied relatively low, there is provided a wireless Internet-based foreign language learning content service method comprising a differential amplification function for differentially reinforcing weak sound compared to strong sound.

Further, preferably, the content file comprises: a) an audio file comprising digital audio data in which the intensity of the sound over time is represented by an integer stream, or b) an audio file comprising text data corresponding to the audio data together with the audio data. It is a text file.

In addition, the engine program further has a function of providing the reproduction speed information of the audio data to the driving unit of the display panel on which the characters are displayed so that the characters displayed on the display panel are inverted in synchronization with the reproduction speed of the audio data.

Hereinafter, an audio signal processing method according to the present invention will be described in more detail with reference to the accompanying drawings.

1 is a block diagram showing a schematic configuration of a system required to service foreign language learning content in a wireless Internet based on the present invention.

The subject providing the service of the foreign language learning content according to the present invention is a content providing system 8, and is mainly composed of a web server 5 and a database server 6. The web server 5 is an Internet web server that is in charge of information exchange with a user who wants to use the service, and requests a service from the mobile communication terminal 2 through the mobile communication service system 3 and the wireless Internet gateway server 4. When receiving a signal to transmit the information stored in the database server 6 to the mobile communication terminal 2 via the wireless Internet gateway server 4 and the mobile communication service system 3 according to a predetermined format. . That is, the web server 5 is located between the wireless internet gateway server 4 and the database server 6 and connects to the mobile communication service system 3 via the wireless internet gateway server 4.

The database server 6 manages many kinds of foreign language learning contents in the form of files, but periodically incorporates and manages new contents. In addition, various general Internet information providing systems such as membership management are required to play a role. Store and manage data.

The wireless Internet gateway server 4 is disposed between the mobile communication service system 3 and the web server 5, and connects to the mobile communication service system 3 via a wireless internet protocol (e.g., a WAP protocol). (5) is connected by wired Internet protocol (e.g., TCP / IP protocol). The wireless internet gateway server 4 converts the encoded request (information) provided from the mobile communication service system 3 to the wireless internet protocol into a request according to the wired internet protocol, and transmits the request to the web server 5. It is a gateway server that converts the response of the web server 5 provided according to the protocol into a response encoded according to the wireless Internet protocol and delivers the response to the mobile communication service system 3.

Information exchange is performed between the mobile communication service system 3 and the mobile communication terminal 2 by a conventional wireless communication method (for example, a CDMA method) as is well known.

2 is a block diagram showing a schematic configuration of a conventional mobile communication terminal. In connection with the application of the present invention, the hardware configuration of the mobile communication terminal 2 is not particularly changed. However, in order to store the downloaded content file, the memory capacity should be large enough. In addition, it is necessary to provide a nonvolatile memory for embedding an engine program for processing a voice signal (to be described later). 2 is a configuration that can meet such a demand.

Functions and functions of the respective components of the mobile communication terminal 2 are well known to those skilled in the art, and therefore, only specific points related to the present invention will be briefly described. The nonvolatile memory 34 stores a downloaded content file or an engine program for processing a voice signal, and EPROM or EEPROM may be a representative example. The key input unit 20 is a means for accessing a content providing site through the wireless Internet, downloading a desired content file, and selecting a content file stored in the nonvolatile memory 34 to set a playback condition to issue a command, and the like. to be. The display panel 28 (mainly an LCD panel) displays graphics or characters to be output under the control of the panel driver 26. ROM 36 contains a basic operating system program. Of course, the voice signal processing engine program may be provided as firmware in the ROM 36 instead of being stored in the nonvolatile memory unit 34. The following description will be given according to the former. The RF unit 32 is a voice by a predetermined process such as demodulation and the like, the radio signal received through the antenna 40 is reproduced to the speaker 24 through the voice signal processing unit 22 and the downloaded content file is a microprocessor unit (MPU) 30 is stored in the nonvolatile memory 34 under the control of the. In addition, the audio output of the microphone 38 is signal-processed according to a predetermined signal method and then wirelessly transmitted. The MPU 30 controls overall operations of each component of the mobile communication terminal and performs necessary calculations.

3 is a software functional block diagram for realizing an audio signal processing method according to the present invention. This is a block diagram showing the function of the engine signal processing engine program, and in practice, such a function is exhibited by the MPU 30. The engine program reproduces or reproduces the audio data read from the nonvolatile memory unit 34 according to the size of the reproduction time adjustment rate α and the differential amplification ratio β set by the user through the key input unit 20. The audio signal processing section 10 includes a playback time adjusting section 12 that adjusts time and a differential amplification section 14 that differentially strengthens the weak sound by amplifying the weak sound by applying a relatively higher gain ratio than the strong sound. And the filter unit 16 which receives the reproduction time and the differential amplified data and performs low pass filtering to remove noise in the high frequency band, and buffers the data processing speed difference between the front and rear ends without interrupting the supply of data to the output stage. It further includes a buffer unit 18 to ensure a smooth supply. In some cases, the key input unit 20 further provides a threshold value x _th of the sound intensity to which the weakening process is applied.

The digital audio data provided to the nonvolatile memory unit 34 expresses the sound intensity with respect to time as an integer stream, and is preferably data normalized by the maximum value. In the case of non-normalized data, the playback time adjusting unit ( 12) or the differential amplifier 14 normalizes in advance before signal processing. The source of digital audio data may be a hard disk or a CD-ROM which stores an audio file (eg, an audio file in wav or avi format), or may be an online input means such as a microphone. In order to process the output signal of the microphone, since the output signal is an analog signal, the process of digitizing it must be preceded.

4 shows an appearance of a mobile communication terminal. Although not shown in the drawings, each of the numeric buttons correspond to a plurality of English characters and Korean characters. The key input unit 20 includes a plurality of numeric keys, up, down, left and right navigation keys, menu keys, confirmation keys, and the like, and the use of the key input unit 20 and its functions are well known to those skilled in the art, and thus descriptions thereof will be omitted.

Next, FIG. 5 is a flowchart illustrating a procedure for downloading a foreign language learning content to a mobile communication terminal by accessing a content providing system using a wireless Internet. In order to learn foreign language listening ability by using the mobile terminal, users of mobile communication terminals must first download and store content files for learning foreign language listening ability. To this end, first, access to a foreign language learning content providing site (hereinafter referred to as the 'content site') through the wireless Internet (S10). Wireless internet access procedures can vary.

7A to 7I exemplarily illustrate screen configurations of a display panel of a mobile communication terminal when downloading and playing content for foreign language learning, which will be described with reference to wireless Internet. When the user selects the 'menu' button of the terminal, a basic menu is displayed on the panel 28 as shown in (a). Selecting a web browser will take you to the Internet connection. As shown in (c), the Internet address may be directly input by using an input key, but it is also possible to select from among pre-stored Internet addresses. In order to find the Internet address conveniently and easily, you may go through the process of displaying various menus as shown in (b). Each menu of (b) is stored with a corresponding Internet address. A wireless Internet connection is made by selecting an internet address from among the listed internet addresses that provides foreign language listening ability learning content.

When the connection is made, check whether the member is using the accessor's telephone number information, and in the case of a non-member, go through the membership registration process (steps S12 and S14). At this time, the web server 5 transmits the voice signal processing engine program to the mobile communication terminal 2 of the new subscriber to be stored in the nonvolatile memory unit 34 (step S16). The route for installing the voice signal processing engine program in the mobile communication terminal 2 may also be provided as firmware through a mobile communication service provider affiliated thereto.

When access is made to the content site, the web server 5 transmits a menu of foreign language learning content to the mobile communication terminal 2, for example, as shown in (e) of FIG. 7 (step S18).

The user manipulates the key input unit 20 among the menus displayed on the panel 28 to select the desired content (for example, selecting CNN No. 1 in FIG. 7F and two of the major news of the day in (g)). When the information is transmitted (step S20), the web server 5 requests and receives the requested content from the database server 6 and sends it to the mobile communication terminal 2 again. Transmit (step S22).

The mobile communication terminal 2 receives the transmitted content file and stores it in the nonvolatile memory unit 24 (step S24).

If the user wants to select more content, the process returns to step S20 again, otherwise the internet connection is terminated (step S28).

By following the above procedure, it is possible to download and store foreign language listening ability learning contents through the wireless Internet using a mobile communication terminal.

Herein, the content file may be configured in various forms. Firstly, an audio file containing only digital audio data expressing sound intensity over time as an integer stream, and secondly, an audio-text file including text data corresponding to the audio data together with the audio data. Of course, a text file including only text data is naturally included, and a file including video data may also be configured. The format of the audio-text file or other specific matters thereof is disclosed in Korean Patent Application No. 2000-29838, titled "Method of Reproducing an Audio Signal Corresponding to Section-Specified Characters and a Reproducing Device." The contents of this application are included in the contents of the present invention.

Next, FIG. 6 is a flowchart illustrating a procedure of reproducing foreign language learning content stored in a mobile communication terminal.

When the user turns on his mobile communication terminal 2 and presses the 'menu' button, for example, a main menu as shown in FIG. 7A is displayed on the panel 28 (step S32).

From the main menu, select '4. If the file playback 'is selected (step S32), various service menus including the' foreign language learning 'menu related to file playback are displayed again on the panel (step S34). In this state, if the 'foreign language learning' menu is selected (step S36), a list of content files related to foreign language learning is displayed. Preferably, the menu and the contents file list related to 'file playback' are fine according to the directory and file management scheme of the computer (step S38).

When the user selects a content file to be played from the list of content files (step S40), a message for setting the playback conditions for the playback speed and the differential amplification rate of the audio data as shown in FIG. Display (step S42). When the user inputs the value, the key input unit 20 transmits the input value to the MPU 30 as a reference when processing the audio data. If the user disregards the play condition input and issues a play command, the normal play speed is played without reinforcement.

When the user issues a playback command (S42), the audio data and the text data are separately processed and output to the speaker 24 and the display panel 28 (steps S44 and S46).

Here, the driving principle of the character data is well known to those skilled in the art, so the description thereof is omitted. However, when the user listens to the playback sound through the earphone to enhance the learning effect, it is also a good way to reverse the character data to match the playback speed of the audio data when displaying on the panel 28 through the character data processing. To this end, the engine signal processing engine program provides the driving unit 26 of the display panel in which the characters are displayed to the display panel 28 by providing the reproduction speed information of the audio data of the character data according to the variation of the reproduction speed of the audio data. The displayed characters are inverted in synchronization with the playback speed of the audio data.

Details of the reproduction processing of audio data will be described later.

In some cases, the user may want to listen to a content file that has been played once (see (I) of FIG. 7), or may want to listen to another content file. In the former case, the process returns to step S42 (step S48), In the latter case, the process returns to step S34 (step S50).

By this process, the downloaded content file can be played.

8 is a flowchart schematically illustrating a method of processing an audio signal applied when reproducing foreign language learning content in a mobile communication terminal.

The user sets the desired reproduction time adjustment rate α and the differential amplification ratio β using the key input unit 20. The reproduction time adjustment rate α is for adjusting the reproduction time of the audio signal longer or shorter than the normal reproduction time, and is determined within the range of α _min ≦ α ≦ α _max . If the value of α is 1, it is reproduced at the same speed as the original normal reproduction speed. If the value is less than 1, the reproduction time is shortened (ie, the reproduction speed is increased) because the amount of the digital signal is reduced. If the value is greater than 1, the playback time increases and the playback sound is slow. In the case of the audio signal, it can be confirmed through actual implementation that α can be reproduced with sufficiently good sound quality even if it is set within the range of 0.7 or more and 3 or less. The differential amplification ratio β is for reinforcing the weak sound of the audio signal relatively more than the strong sound, and is determined within the range of β _min ≦ β ≦ β _max . In the case of the audio signal, β also has a value within the range of 1 to 3, it can be confirmed through the actual implementation that it can be reproduced with a sufficiently good sound quality.

The MPU 30 receives the reproduction time adjustment rate α and the differential amplification ratio β input by the user, and provides them to the reproduction time adjustment unit 12 and the differential amplification unit 14 (step S110). When these values are provided during the processing of the audio signal, the reproduction time adjustment section 12 and the differential amplifier section 14 are interrupted, and the existing values are replaced by the newly provided reproduction time adjustment rate α and the differential amplification ratio β.

The nonvolatile memory 34 stores the original digital audio signal x (·) before processing, which is read by the signal processing section 10 (step S112). The unit of reading signal is frame. The frame includes N sample data. In order to adjust the playback time, the next frame is not read from the end of the previous frame, but adjacent frames are read in duplicate. That is, the next frame F _m reads out a predetermined number of samples repeatedly between adjacent frames by reading from the sample of the position where the number of samples less than N is skipped from the start position of the previous frame F _m-1 . A more detailed description thereof will be described later.

For each frame read through the data reading step (step S112), the signal processing unit 10 performs a differential amplification step (step S14) and a time axis conversion step (step S116), that is, a playback time conversion step. The differential amplification step (step S114) and the time base conversion step (step S116) may process the audio signal in the order of processing the differential amplification first and then processing the time base conversion as shown in the flowchart of FIG. 8, but the reverse order is also possible.

The differential amplification step (step S114) is a step of amplifying by substituting the sample data x _i of each frame into a predetermined differential amplification function f (x _i ) corresponding to the differential amplification factor β. Here, the differential amplification function f (x _i ) is a function in which the amplification gain is relatively low, especially from weak to strong. Therefore, by converting the audio data using this function, it is possible to differentially strengthen the weak sound compared to the strong sound.

In the time-base conversion step (step S116), the next frame F _m may be a memory (which may be physically a specific area of the nonvolatile memory 34 or may be an internal memory of the signal processing unit 10, that is, the MPU 30). Is added so as to overlap the previous frame F _m-1 already recorded as the time-axis conversion signal y (占) by a predetermined number of samples. Here, the sample of the duplicated part is weighted and added, and the non-duplicate part is added as it is. In addition, the number of samples of the overlapping portion is added in a manner of increasing or decreasing corresponding to the size of the reproduction time adjustment rate?. Therefore, the original signal x (·) has its time axis length converted in proportion to the reproduction time adjustment rate α.

In this way, the audio data converted into the reproduction time adjustment rate and the differential amplification rate set by the user by the differential amplification and time-base conversion processing is filtered by the filter unit 16 and recorded in the buffer unit 18 (step S118). . Undesired noise can be introduced during the differential amplification and time-base conversion process. Since the noise usually has a high frequency component, the low pass filtering process removes the noise. Filtering may be performed in units of frames, but for efficient processing, it is preferable to process a plurality of frames constituting one packet in units of processing. The filtered packet data is recorded in the output buffer 18. It is preferable to provide a plurality of output buffers 18 to form a ring buffer for real time reproduction of audio data. The signals processed through the time-base conversion step and the differential amplification step are written to the plurality of output buffers 18, but the output buffers for writing the packet data are cyclically replaced. For example, assuming that there are four output buffers Buf_1, Buf_2, Buf_3, and Buf_4, the order of writing the packet data is circulated in the order of 'Buf_1-> Buf_2-> Buf_3-> Buf_4-> Buf_1 ....'.

In parallel with the writing of the packet data to the output buffer 18, the packet data recorded in the plurality of output buffers is first outputted to the voice signal processor 22 in the packet data of the output buffer prior to the recording order (step S120). . That is, the cyclic order of the output buffers is also the same as the above write order of packet data. The digital data input to the audio signal processing unit 22 is converted into an analog signal and provided to the speaker 24 through post-processing necessary for audio output such as amplification processing.

9 is a flowchart showing an algorithm of a time axis conversion step (step S16). FIG. 10 is a reference diagram for explaining the principle of obtaining the time-axis conversion signal y (·) by increasing or decreasing the length of the original audio signal x (·) corresponding to the magnitude of the reproduction time adjustment rate α set by the user. . For ease of understanding in FIG. 10, the frame Fm includes 320 samples, the reproduction time adjustment rate α is 2, and the basic difference value Sa of the starting point between each frame of the original signal x (·) is 120 (sample). The case where the check range (window) k _max for detecting the optimum correlation on the basis of the similarity of the waveform between the signal x (·) and the time-axis conversion signal y (·) is ± 40 (sample) based on the Sa value is taken as an example. . These values are exemplary values and can be changed to other values according to the application environment.

First, the first frame F0 of the original audio signal x (·) is read from the nonvolatile memory 34, and the time axis conversion signal is transferred to the second memory (another area of the nonvolatile memory 34 or the internal memory of the MPU 30). The copy is carried out as is (y) (step S122).

Then, the value of the frame index m is set to 1 (step S124).

Thereafter, the next loop is executed until the conversion of the original signal x (占) is completed (step S136).

First, the next frame F1 should be read from the original signal x (·) and added to the time axis converted signal y (·). Here, the start position when the next frame F1 is read from the original signal x (·) is variably determined. The criterion is determined by the synchronized lag K1 with the previous frame F0 copied in the time-axis conversion signal y (·). Also, the position to be read out and added as the time axis conversion signal y (·) of the frame F1 also varies depending on the size of the reproduction time adjustment rate?.

This is generally described as follows. The synchronization delay Km generally means a change value of the start position of the next frame of the original signal x (·) so as to have an optimal correlation with the time-axis conversion signal y (·). In order to calculate the synchronization delay Km, first, the synchronization delay Km is obtained using the following equation for the range between the lower limit mSa-40 and the upper limit mSa + 40 of the correlation check range. The synchronization delay K _m is used to determine, within a predetermined range, that the next frame F _m has an optimum correlation with the previous frame F _m-1 already recorded as the time-base conversion signal y (·).

<Equation 1>

<Formula 2>

However, L means the number of samples of the overlap part between adjacent frames.

Using the two equations above the ground determined the Km synchronization delay with the best correlation, reads the next frame F _m, including N samples, using the values from the original signal x (and) a. The read start position of the next frame F _m is S _a ± K _m, which is the value obtained by subtracting the synchronization delay K _m from the frame start interval S _a at the read start position of the previous frame F _m-1 , where 0 <S _a ± K _m < N) samples will be skipped. For example, referring to FIG. 10, when K ₁ , K ₂ , and K ₃ are set to 20, −10, and 35, respectively, the read start positions of the second, third, and fourth frames F1, F2, and F3 are 140, respectively. It is the 230th and 395th samples. Of course, the number of samples in each frame is always 320, which is N. When the original signal x (·) is read in this manner, a significant amount of samples are duplicated in the previous frame and the next frame. This overlapping portion decreases (in the case of α> 1) or increases (in the case of α <1) in accordance with the magnitude of the reproduction time adjustment rate α in the time-axis conversion signal y (·). Note that the reading start point of each frame does not vary regularly by the product of the basic difference value Sa and the frame index m of the starting point between each frame, but irregularly according to the size of the optimal correlation Km determined using Equations 1 and 2 Is the point.

After this reading, the frame F _m is added to the time-axis conversion signal y (·). The start position of the next frame F _m added to the time-axis conversion signal y (·) is determined by the frame start interval S _a , the reproduction time adjustment rate α, and the product mαS _a of the index m of each frame. Therefore, in FIG. 4, since αSa = 2x120 = 240, additional positions of the second, third and fourth frames F1, F2, and F3 are 240, 480, and 720. In addition, the front part of the next frame F _m overlaps with the back part of the previous frame F _m-1 . The overlapping portions of both frames are weighted using Equations 3 and 4 below, and the remaining portions of the next non-overlapping frame F _m are copied as they are.

<Equation 3>

<Equation 4>

Where g (j) is a weight function. This weight function g (j) is obviously a linear function, but an exponential function may be applied.

In this manner, the original signal x (·) is read in units of frames and added to the time axis conversion signal y (·). In this process, each time one frame is processed, the value of the frame index m is increased by one (step S130). Then, it is checked whether the size of the frame index m is a multiple of the number of frames PL constituting one packet. When the frame index m reaches a multiple of PL, the data amount of the time-base conversion signal y (·) has reached the packet data amount. In this case, the converted data is subjected to low pass filtering and then recorded in the buffer unit 18 ( Steps S132 and S134).

Then, the loop composed of steps S126 to S136 is repeatedly executed until all data of the original signal x (·) is processed while checking whether the original signal x (·) has been converted to the end. In this manner, time axis conversion processing according to the reproduction time adjustment rate α set by the user is performed. FIG. 13A exemplarily shows a waveform diagram of a time of a normalized original signal x (·). This waveform is a waveform diagram of the time of the signal y (·) obtained by performing time-base conversion so that its length is doubled (that is, when the reproduction time adjustment ratio? Is 2). Comparing these waveforms, it can be seen that the original signal x (·) is approximately doubled on the time axis, but its morphological characteristics remain almost intact, and a good sound quality can be felt by actually listening to the reproduction sound.

Next, FIG. 11 is a flowchart illustrating an algorithm for differentially amplifying by applying amplification higher than weak sound.

The differential amplification algorithm handles samples. Therefore, this algorithm is characterized in that it can be executed in parallel with the timebase conversion algorithm of FIG. For example, in the time-base conversion algorithm, it is possible to read the original signal x (·) in units of frames, perform a differential amplification algorithm on each sample of the frame, and then convert it into a time-base conversion signal y (·). That's the way. As another selectable method, when the time-axis conversion signal y (·) is gathered by the packet data amount, the low pass filtering process is performed. Before the low-pass filtering, the differential amplification of the sample unit for the time-axis conversion signal y (·) is executed. That's the way. Of course, it is impossible to perform differential amplification after low pass filtering, so it is effective to put differential amplification ahead of low pass filtering because low pass filtering has the purpose of removing noise.

The differential amplification algorithm will be described in detail with reference to FIG. 11 as follows.

The differential amplification method can be modified in various forms depending on which function is applied as the differential amplification function and whether the differential amplification object is to be distinguished according to the value of the sample data. Basically, the differential amplification function f (x _i ) should be selected from among functions in which the amplification gain is applied relatively low from the weak sound to the strong sound. A function that satisfies this requirement is a monotonically increasing function, for example, a linear function with f (x) = x or an exponential function with f (x) = x ^{1 / β} , as illustrated in FIG. . Substituting the sample data into the variables of these functions and taking the value of the function in place of the sample data, the weakness is strengthened differentially compared to the strong sound.

(B) of Figure 12 if only the sample data of a larger value than a value optionally differential amplifier based on the threshold value x _th and apply the methods to the sample data below a threshold value x _th retains its original value as it is It is shown.

FIG. 12 (c) shows that the sample data having a value larger than this value based on the threshold value x _th is used to calculate a function value using the differential amplification function f (x _i ) = x _i ^{1 / β} and the value smaller than the threshold value. The sample data having a value shows a case of applying a method of replacing the original sample data x _i by calculating a function value using a differential amplification function f (x _i ) = x _i ^β .

Of course, instead of going through the procedure of comparing the sample data x _i with the threshold value x _th , it is also possible to calculate all the sample data as a differential amplification function and convert it into the function value.

11 is a flowchart of an execution algorithm in the case of performing differential amplification according to FIG. 12C among these methods.

Referring to this flowchart, first, the size of the sample to be processed x _i is compared with a threshold value x _th (step S140). This threshold x _th can be set by the user or predefined by the system.

As a result of the comparison, when the sample data x _i is larger than the threshold value x _th , the sample data x _i is replaced by the function value x _i ^{1 / β} , and conversely, when the sample data x _i is smaller than the threshold value x _th , the sample data x _{i is used.} Is replaced by the function value x _i ^β (steps S142 and S144). If x _i is smaller than the threshold x _th , applying the exponential function to the differential amplification function as in step S142 has the advantage that the noise due to the small negative value can be effectively attenuated. On the other hand, if the linear function is applied as the differential amplification function as in the case of FIG. 12 (b), the amount of calculation is relatively small. If the original signal x (·) was recorded in a noise-free space such as a recording studio, it would not be necessary to apply an exponential function if x _i was less than the threshold value x _th .

After each sample data is processed as above, it is checked whether the sample index i exceeds an integer multiple of the sample number PS of the packet data (step S146), and if the number of processed sample data has not yet reached one packet, the sample index Only i is incremented by 1 (step S148), and the operation using the new differential amplification function is repeated for the next sample data.

When the number of processed sample data reaches the packet data amount, the low-pass filtering process for noise reduction is performed on the differentially amplified data set {x _i } (step S150). The filtered data is then recorded in the buffer unit 18, and finally, the analog signal is converted through the voice signal processing unit 22 and output to the speaker 24 (step S152). Of course, when the differential amplification algorithm is arranged as part of the time-base transformation algorithm, the calculation step of the sample data using the differential amplification function is only incorporated.

This processing routine is repeated repeatedly until all the sample data of the original signal is processed (steps S154 and S148).

FIG. 13C is a waveform diagram of a signal obtained by actually differentially amplifying the original signal x (·) of FIG. 13A by applying a predetermined differential amplification factor. By comparing these two waveform diagrams, it can be seen that the low amplitude parts, rather than the loud intensity parts, are relatively more amplified. This is consistent with the intended result.

The present invention provides a foreign language to a person who accesses the site through the wireless Internet using a mobile communication terminal capable of wireless internet access, such as cellular, personal mobile communication (PCS), and international mobile telecommunications-2000 (IMT-2000). It provides contents for listening ability learning, and also provides a voice signal processing engine program that reinforces the playback speed and weakness of audio data.

The method according to the invention makes it very easy to learn the listening ability of a foreign language. There is no longer any temporal and spatial constraints in learning foreign language listening skills. In particular, during the rush hour, you can learn to listen to a foreign language using a mobile communication terminal in the subway, bus, or car, so that you can utilize your time efficiently.

In addition, since only one mobile communication terminal can learn foreign language listening ability, it is no longer necessary to carry a cassette tape record or a cassette tape in addition to the mobile communication terminal.

From a user's point of view, foreign language learning will be possible at a lower cost.

In addition, the user can adjust the playback speed of the voice data according to the level of his or her listening ability can be greatly enhanced by listening to the enhanced part can not greatly hear.

Although the present invention has been described above according to an embodiment of the present invention, it will be apparent to those skilled in the art that various modifications may be made without departing from the spirit of the present invention.

Claims (11)

After accessing the site through the wireless Internet using a mobile communication terminal, the member authentication process is performed, and if the member is not a member, the member registration process is performed, and then the voice signal processing engine program is automatically installed and installed in the mobile communication terminal. Making; And Transmitting the list information of the foreign language listening ability learning contents that can be provided to the mobile communication terminal which has been authenticated by the member, and transmitting the selected content file to the mobile communication terminal and storing the content file, wherein the playback speed and the differential amplification rate set by the user are set; According to the method for simultaneously processing the adjustment of the playback speed of the audio data included in the content file and the enhancement of the sound by the differential amplification, The voice signal processing engine program, When a user selects a specific content file from among the foreign language listening ability learning contents stored in the mobile communication terminal and gives a playback command, the reproduction time adjustment rate α specifying the reproduction speed of the audio data and the differential amplification rate β for the strong sound of the weak sound Ask the function to receive the setting value; The audio data x (·) is continuously read from the first memory in units of N samples, and the next frame F _m crosses a smaller number of samples than the N number at the start of the previous frame F _m-1 . A data reading function for reading a predetermined number of samples repeatedly between adjacent frames by reading from the sample of the jumped position; The next frame F _{m is added} to the second memory so as to overlap with the previous frame F _m-1 already recorded as the time-base conversion signal y (·) by a predetermined number of samples, and the samples of the overlapped portion are weighted, added, and non-overlapping. The portion is copied as it is, and the number of samples of the overlapping portion is added in such a manner as to increase or decrease in correspondence with the size of the reproduction time adjustment rate α so as to obtain a signal y (·) whose time axis length is converted in proportion to the reproduction time adjustment rate α. Time axis conversion function; And Sample data x _i of each frame is amplified by substituting a predetermined differential amplification function f (x _i ) corresponding to the differential amplification ratio β, in particular, and the differential amplification function f (x _i ) is amplified from weak to strong. A wireless internet-based content learning method for learning a foreign language, comprising a differential amplification function for differentially enhancing a weak sound compared to a strong sound since a function of applying a gain is relatively low. delete 2. The audio file of claim 1, wherein the content file comprises: a) an audio file comprising digital audio data representing a sound intensity over time as an integer stream, or b) text data corresponding to the audio data together with the audio data. The engine program is a text file, and the engine program provides information on the reproduction speed of the audio data to a driving unit of the display panel on which the characters are displayed so that the characters displayed on the display panel are inverted in synchronization with the reproduction speed of the audio data. Content service method for learning foreign languages. The method of claim 1, wherein the engine program comprises: a plurality of output buffers, and any one of the plurality of output buffers by binding a plurality of frames of a signal processed through the time-base transformation and the differential amplification into one packet. In particular, the output buffer for writing the packet data is cyclically replaced, and in parallel with this, the packet data recorded in the plurality of output buffers has the audio signal including the speaker first. Wireless Internet-based foreign language learning content service method characterized in that it further comprises a data output function for controlling to be output to the processing unit. The method of claim 1, wherein the engine program further comprises a filtering function for removing noise by performing low pass filtering on the differentially amplified sample data for each predetermined data amount. The method of claim 1, wherein the engine program is started with respect to the data read out function reading of the next frame F _m where the acceleration a synchronous delay K _m for the starting frame in the reading start position of the previous frame F _m-1, the interval S _a A position is a position skipped S _a ± K _m (where 0 <S _a ± K _m <N) samples, and the synchronization delay K _m is within a predetermined range, where the next frame F _m is the time-axis conversion signal y A wireless Internet-based content learning method for learning a foreign language, characterized in that it is determined to have an optimal correlation with the previous frame F _m-1 already recorded as (·). The method of claim 1, wherein the time-base conversion the time-base conversion start position of each frame is added to the signal y (and) is the product of the frame time interval, S _a, the playback time update ratio α, and the index of each of the frames m mαS _a Wireless Internet-based foreign language learning content service method, characterized in that determined by. The method of claim 1, wherein the differential amplification function f (x _i ) is a monotonically increasing function. The method of claim 8, wherein the differential amplification function f (x _i ) is a function expressed as f (x _i ) = x _i ^{1 / β} . The wireless internet-based foreign language learning content service according to claim 1, wherein the differential amplification operation using the differential amplification function f (x _i ) is performed for each frame read in the data reading procedure prior to the time-base conversion process. Way. The method of claim 1, wherein the differential amplification operation using the differential amplification function f (x _i ) is performed on the time axis transform signal y (·) obtained after the time axis transform processing. .