KR100541605B1 - Method for storing audio and multimedia data including additional information in digital audio broadcasting receiver system - Google Patents

Abstract

The present invention relates to a method for storing sound sources and multimedia data including additional information in a digital audio broadcasting system. More particularly, the present invention relates to a method of directly acquiring and storing MPEG-type audio data from a channel decoding chip and using characteristics of MPEG audio. A method of generating and storing data is provided.

In the digital audio broadcasting system of the present invention, a sound source and multimedia data storage method including additional information may include a first step of receiving a digital broadcast signal and outputting a broadcast signal of a frequency band selected by a user; A second step of decoding and outputting the outputted broadcast signal into a data bit stream of a type different from an MPEG-type digital audio bit stream; Extending the decoded MPEG audio bit stream into original signal audio data and processing other data bit streams; Technical features include a fourth step of generating additional information data and inserting it into an original signal and a fifth step of storing the signal into which the additional information data is inserted.

Accordingly, in the digital audio broadcasting system of the present invention, a method for storing sound sources and multimedia data including additional information is to reduce the load on the system and to store and reproduce the digital broadcasting data on the user's device faster while using the storage medium more efficiently. It is possible to reproduce the signal of the same sound quality as the original signal repeatedly during playback.In case of acquiring and storing the original signal, it is possible to insert and store the user's classification information so that various information can be easily retrieved later. It is possible to replay the information and to grasp the information briefly without directly accessing the stored information.

DAB, storage medium, original signal, additional area

Description

Method for storing audio and multimedia data including additional information in digital audio broadcasting receiver system}             

1 is a block diagram showing a conventional digital audio broadcasting receiver.

2 is a block diagram schematically illustrating a digital broadcasting receiver capable of storing a digital broadcasting signal in the form of an MPEG according to an embodiment of the present invention.

3 is a block diagram illustrating a process of encoding an audio signal according to the present invention.

4 is a block diagram illustrating a process of decoding an audio signal according to the present invention.

5A is a frame structure of MPEG audio data according to the present invention;

      (B) is a frame structure of digital audio broadcasting audio data according to the present invention.

6 is a block diagram showing a structure in which ID3-Tag is added to an MP3 file.

7 is a block diagram showing the structure of ID3-Tag V2.4.0.

           <Description of the symbols for the main parts of the drawings>

      200. Receiving unit 210. Channel decoding unit

      220. Audio decoding unit 230. Data processing unit

      240. Control unit 250. Storage unit

      260. Frame Packing 270. Unpacking the Frame

      280. Rebuilding 290. Subbands

The present invention relates to a method of storing sound sources and multimedia data including additional information in a digital audio broadcasting system. More particularly, the present invention relates to a method of directly acquiring and storing MPEG-type audio data from a channel decoding chip and characteristics of MPEG audio. The present invention relates to a method for generating and storing data appropriately.

As we enter the 21st century information society, the age of full-scale broadband multimedia service has come along with the digitization of various wired and wireless communication systems such as mobile communication, satellite communication, and high-speed subscriber lines. However, broadcasting and telecommunications fields such as television and radio, which are the media that can easily access information such as audio and video in everyday life, are relatively less converted to digital systems than other fields.

Especially in radio broadcasting, analog broadcasting such as AM (Amplitude Modulation) and FM (Frequency Modulation) has provided a variety of information to many people in the last 20 centuries. Power efficiency is reduced by using high transmit power, and there are technical limitations such as spectral efficiency being degraded by using different frequencies in the near region to avoid co-channel interference.

In order to overcome the limitations of the conventional analog radio broadcasting system and provide various multimedia broadcasting services, researches on various types of digital audio broadcasting systems (DAB: Digital Audio Broadcating) have been conducted. have. Looking at the representative digital audio broadcasting system that is currently being developed or implemented, the representative examples are the In-Band-On-Channel (IBOC) method in the US and the Eureka 147 method in Europe.

The IBOC system of the United States is being developed to provide an enhanced service that can reproduce the analog broadcast coverage (Coverage) as it is, by using the existing FM radio band as a digital broadcast service at the same time. The system has an Orthogonal Frequency Division Multiplexing structure using bilateral bands separated by a certain frequency from the center of the analog radio service channel, and audio is compressed by MPEG-2 AAC to provide stereo CD level sound quality. do.

The IBOC system has two operation modes, a hybrid mode and an all-digital mode, depending on the transmission mode. The hybrid IBOC is a system that simultaneously transmits a digital signal together with an existing analog signal on the same channel. Used during the switching period, the all-digital IBOC transmits only digital signals in the FM band.

The European Eureka 147 system has a strength in multipath propagation by applying Coded Orthogonal Frequency Division Multiplexing using multiple carriers, and complements it by using a guard interval, thereby implementing a single frequency broadcasting network. In addition to the advantages, there is also a possible mobile reception service. It can operate in the VHF, UHF and L / S bands from 30MHz to 3GHz, and provides audio and data services with 1.536MHz broadband transmission. However, due to the broadband transmission as described above, implementation within the FM broadcast frequency band is impossible without interrupting the existing analog service, and thus has a disadvantage in that a new frequency band other than the FM band is required.

Digital audio broadcasting as described above is moving beyond simple audio services such as AM and FM, and is oriented toward mobile multimedia services that can accommodate data broadcasting and video.

1 is a block diagram showing a conventional digital audio broadcasting receiver. When the digital audio broadcast signal is input to the receiver in the microcontroller 100 through the antenna, the receiver outputs only the baseband signal of the frequency band selected by the user to the channel decoder 110 by the receiver control signal of the controller.

The channel decoding unit 110 decodes the corresponding baseband signal into a data bit stream of a different type from that of the MPEG audio bit stream.

The MPEG layer2 audio bit stream decoded by the channel decoder 110 is extended to the original audio data by the MPEG audio decoder 120 and an analog signal is transmitted through the digital / analog converter 130. After the conversion to the output through the amplifier to the speaker 140.

In the prior art as described above, when the audio data is to be stored in a receiver or the like, data converted into an analog signal through the digital-to-analog converter 130 is stored in a storage device such as a memory. That is, the storage method of the audio data developed so far is a method of simply storing the audio data of the analog signal as an analog medium or digitizing the analog signal as described above.

Although the method of storing the analog airwave broadcasting on a magnetic recording medium or another medium is very simple, the problem of limitation of the storage capacity and the deterioration of sound quality occurs. Similarly, there was a problem that deterioration of sound quality occurs. That is, the conventional methods of storing or digitizing the analog signal to an analog medium have a problem in that the stored analog signal cannot obtain the same sound quality as the original signal, but also degrades the sound quality.

Accordingly, the present invention is to solve the above disadvantages and problems of the prior art, a new data structure by storing the same digital signal as the original signal in the digital broadcasting system, by generating various additional areas in the data stored in the receiver To provide a way to store the original signal.

The first object of the present invention is to receive a digital broadcast signal and output a broadcast signal of a frequency band selected by a user; A second step of decoding and outputting the outputted broadcast signal into a data bit stream of a type different from an MPEG-type digital audio bit stream; Extending the decoded MPEG audio bit stream into original signal audio data and processing other data bit streams; A digital audio broadcasting system comprising a fourth step of generating additional information data and inserting the additional information data into an original signal and a fifth step of storing the signal into which the additional information data is inserted are achieved by a sound source and multimedia data storage method including additional information. .

Details of the above object and technical configuration of the present invention and the effects thereof according to the present invention will be more clearly understood by the following detailed description with reference to the drawings showing preferred embodiments of the present invention.

2 is a block diagram schematically illustrating a digital broadcasting receiver capable of storing a digital broadcasting signal in the form of an MPEG according to an embodiment of the present invention. As shown in FIG. 2, the digital broadcast receiver includes a receiver 200, a channel decoder 210, an audio decoder 220, a data processor 230, a controller 240, and a storage 250. do.

When the digital audio broadcast signal is input through the antenna, the reception unit 200 outputs only the baseband signal of the frequency band selected by the user to the channel decoding unit 210 by the control unit reception signal of the control unit.

The channel decoder 210 decodes the corresponding baseband signal into a data bit stream of a different type from the MPEG audio bit stream.

The MPEG layer 2 audio bit stream decoded by the channel decoder 210 is extended to the original audio data by the audio decoder 220, and the data processor 230 processes other data bit streams.

The control unit 240 allows the receiver to receive a broadcast signal of a frequency band selected by the user, controls the decoding process of the channel decoder, analyzes the received signal, classifies various signals, generates additional information data, and the like. Generates data inserted in the original signal.

The storage unit 250 stores the MPEG audio processed by the audio decoding unit 220 and the data processing unit 230. The service data bit stream decoded by the channel decoder 210 is appropriately processed to output the service data to the display. The storage unit 250 stores and records received data and various driving programs, and the storage medium includes a flash memory, a hard disk, a floppy disk, a CD-DA, a CD-ROM, a DVD-ROM, a memory stick, and the like.

3 is a block diagram illustrating a process of encoding an audio signal according to the present invention. Referring to FIG. 3, when PCM audio samples are input to an encoder, the size of a signal in each subband is obtained by passing through 32 equally divided filter banks in a frequency domain. The input samples are transformed into the frequency domain through the FFT, and then a shielding threshold for 32 subbands is obtained using a psychoacoustic model. A subband with a large shielding threshold may be shielded even if a lot of quantization noise is generated by allocating fewer bits, but in the case of a subband with a small shielding threshold, a large number of bits should be allocated to reduce the occurrence of quantization noise.

The bit allocation operation required to match the measurements of the non-transmitted subbands to the standard format of the PCM audio signal is performed. The bit allocation determines the total number of bits available when the rate is set, so all bits are allocated to each subband. Continue until you do. If the bit allocation is insufficient, the quantization noise is not shielded and the sound quality is reduced. If a large number of bits are allocated, the transmission rate is high and the compression rate is reduced. After the bit allocation operation, the fixed bit rate is adjusted.

In addition, the measured value of the untransmitted subband is recalculated and coded together with the scale factor selection information. When the PCM audio signal is quantized into the arithmetic values and coded into subband samples, the audio signal is finally made into a sound source file, thereby producing an audio signal suitable for the audio format. The subsampled audio signal is made of coding symbols in a quantizer and a coding block, and a frame packing block is a real audio bit stream and additional information and a cyclic redundancy check (CRC) or Packing is performed by combining PADs (Program Associated Data) which are closely related to the coded audio signal. The digital audio broadcast audio frame is generated through the packing block.

4 is a block diagram illustrating a process of decoding an audio signal according to the present invention. FIG. 4 illustrates a process of re-decoding the digital audio broadcast audio frame generated by the encoding process of FIG. 3 and converting the digital audio broadcast audio frame to a PCM audio sample as an original signal. The packed digital audio broadcast audio frame 260 is a frame of a decoder. The data is separated into PADs that are in close association with the actual audio bit stream and additional information, CRC and coded audio signals as they pass through the Unpacking 270 block. Relevant data of the PAD is separately processed and the audio bit stream and additional information data are converted into subband 290 samples via a reconstruction 280 block.

Data to be obtained by the user receiver of the present invention is MPEG audio information, not a wave signal or a recompressed signal, which is an output of a conventional decoder. This is a method of acquiring the compressed audio data with a high level of sound quality that is hardly different from the original sound because the processing load of hardware is reduced when the acquired data is to be reproduced and stored. do.

FIG. 5A is a frame structure of MPEG audio data according to the present invention, and FIG. 5B is a frame structure of digital audio broadcast audio data according to the present invention. The digital audio broadcasting audio coding system uses MPEG audio layer 2. Referring to FIG. 5A, a data frame represents a decodeable MPEG-type bit stream. It typically has 1152 samples, each frame contains header, CRC, bit allocation, Scale Factor Select Information (SCFSI), Scale Factor, subband, sample and auxiliary data. (ancillary data). The digital audio broadcast audio data frame of Fig. 5B shows a bit stream of the digital audio broadcast audio frame. Each frame is composed of header, CRC, bit allocation, scale factor selection information, scale factor, subband, sample, stuff, X-PAD, ScF-CRC, F-PAD, and so on.

When the MPEG audio data frame of FIG. 5A and the digital audio broadcasting data frame of FIG. 5B are compared with each other, a header, a CRC, a bit allocation, a scale factor selection information, a scale factor, a subband sample, and the like are commonly provided. It can be seen that. In addition, the MPEG audio of FIG. 5A includes auxiliary data, but there is a difference in that the auxiliary data does not exist in the digital audio broadcasting audio of FIG. 5B. The auxiliary data may be configured by storing arbitrary data so as to conform to a standard in digital audio broadcasting. In this case, you need to modify the header.

The header consists of a fixed field of 32 bits and has information such as layer information, frequency sampling, and data rate, and the CRC uses an error check bit using a cyclic binary code to detect an error occurring during data transmission. to be. Since the cyclic code is used, error detection performance is high and an encoder or decoder (error detection circuit) is easily realized. ITU-T Recommendation G.704 specifies CRC-6, CRC-4 and CRC-5 for error detection in 1.544, 2.048 and 6.312 Mbps multiplexers, respectively. Where the ^{CRC-6 = X 6 + X} + 1, CRC-4 = X 4 + X + 1, CRC-5 = X 5 + X 4 + X 2 +1.

The bit allocation is to independently adjust the bit allocation for each subband of each frame, effectively using the shielding effect, and completely discarding information about a band including only signals of a level not recognized by the auditory system as a result of the shielding. In this case, no bits are assigned to the band.

The scale factor selection information is a 2-bit code in which many scale factors are indicated for each subband encoded in an audio frame, and the scale factor is a frequency band called a scale factor band without performing quantization on the entire audio band. The band-level level is expressed as a scale factor value. Data of the same scale band is multiplied by the same scale factor value to obtain the original signal by scaling the actual frequency domain data. The scale factor is divided into bands and a large portion of the encoder is processed into a group of scale factor bands. Since the width of the scale factor is made by mimicking the critical band of the human auditory system, the scale factor has a different rental width with respect to the sampling frequency and the transform length.

6 is a block diagram showing a structure in which ID3-Tag is added to an MP3 file. As shown in FIG. 6, the ID3v1.1 Tag is located at the end of the MP3 file, and its size is fixed to 128 bytes. The ID3v1.1 Tag starts with the string 'TAG'. It has 30 bytes for the title, 30 bytes for the artist, 30 bytes for the album, 4 bytes for the year, 29 bytes for the comment, 1 byte for the track, and 1 byte for the genre.

7 is a block diagram showing the structure of ID3-Tag V2.4.0. As shown in FIG. 7, the ID3v2 Tag is located at the beginning of the MP3 file, and the ID3v2 tag starts as a character string 'ID3' like the ID3v1 Tag.

The structure of ID3v2 Tag consists of header, extension header, frame, padding and footer. The header portion has 3 bytes of ID3, 2 bytes of version, 1 byte of flags, and 4 bytes of size. Here, the size of the entire ID3v2 Tag can be obtained from the size.

The extended header, padding, and footer are optional, and the frame includes information such as title, artist, album, and the like of the ID3v1 tag. The frame ID is 4 bytes, the frame size is 4 bytes, the frame flag is 2 bytes, and the frame body is the frame size byte that can be obtained from the frame size.

The method for acquiring additional information automatically generates ID3-Tags when storing digital audio broadcasts in the MPEG format, inserts them into the MPEG data, and stores them together. The ID3-Tag may include various contents such as a track, a title, an artist, an album, a year, a comment, a genre, and the like. The FIC (Fast) having information on a broadcast standard and a program type in a digital audio broadcast signal is transmitted. It shows a structure similar to SI (Service Information) in information channel (Data) content. Although the ID3-Tag does not include additional character information, although MP3 (MPEG audio layer-3) is widely used, 128-bit additional data is added.

ID3v2 can store various data as a container format. Therefore, when multimedia data transmission including a sound source and a still image is implemented, various data can be stored in an MP3 file.

The following relates to another embodiment. Digital audio broadcasting has an advantage that can provide a variety of multimedia content. Hereinafter, an embodiment will be described in which the digital audio broadcasting receiver can purchase the multimedia content and play it on an external device.

The digital audio broadcasting receiver according to the above embodiment may analyze a user interface unit, an external interface unit using a wired or wireless network, parse a received digital broadcasting signal, and purchase purchase information of the multimedia content added to the data stream or the specified specific multimedia content. Control to display on the screen the extracted purchase information supplied from the authentication processing unit and the parsing unit performing authentication related to the purchase, and controls the authentication processing unit to perform the authentication according to the user's purchase instructions through the user interface And a system controller for controlling the extracted and authenticated multimedia content data to be transmitted to an external device using an external interface. The multimedia content includes at least one audio data, the authentication of which includes a payment or descrambling function of the content data.

In addition, the external device may be a portable player, and may be an audio player or a personal information processor capable of processing audio / video. The digital broadcast receiver capable of purchasing the multimedia contents enables the user to play and purchase the multimedia contents without the limitation of time and space through the portable playback device at a convenient and low cost for the general user.

Although the present invention has been shown and described with reference to the preferred embodiments as described above, it is not limited to the above-described embodiments and those skilled in the art without departing from the spirit of the present invention. Various changes and modifications will be possible.

Accordingly, in the digital audio broadcasting system of the present invention, a method for storing sound sources and multimedia data including additional information is to reduce the load on the system and to store and reproduce the digital broadcasting data on the user's device faster while using the storage medium more efficiently. It is possible to reproduce the signal of the same sound quality as the original signal repeatedly during playback.In case of acquiring and storing the original signal, it is possible to insert and store the user's classification information so that various information can be easily retrieved later. It is possible to reproduce the information and to briefly grasp the information without directly accessing the stored information. When the addition of additional data becomes practical, various multimedia services are possible.

Claims (5)

In a digital audio broadcasting system, a sound source and multimedia data storage method including additional information, Receiving a digital broadcast signal and outputting a broadcast signal of a frequency band selected by a user; A second step of decoding and outputting the outputted broadcast signal into a data bit stream of a type different from an MPEG-type digital audio bit stream; Extending the decoded MPEG audio bit stream into original signal audio data and processing other data bit streams; A fourth step of generating additional information data and inserting the additional information data into an original signal; And A fifth step of storing a signal into which the additional information data is inserted; Sound source and multimedia data storage method including the additional information in the digital audio broadcasting system, characterized in that comprises a. The method of claim 1, The method of generating the additional information data of the fourth step comprises storing ID3-Tag when storing the digital audio broadcast in the MPEG format, and storing the sound source and the multimedia data including the additional information in the digital audio broadcasting system. The method of claim 2, The ID3-Tag includes at least one of a track, a title, an artist, an album, a year, a comment, and a genre of the sound source and multimedia data storage method including additional information in the digital audio broadcasting system. The method of claim 2, The ID3-Tag is a sound source and multimedia data storage method including the additional information in the digital audio broadcasting system, characterized in that the additional data of 128 bits size is added. The method of claim 1, The storage medium of the fifth step may include any one of flash memory, hard disk, floppy disk, CD-DA, CD-ROM, DVD-ROM, and memory stick. Sound source and multimedia data storage method.

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