JPH0685703A - Receiver - Google Patents

Abstract

PURPOSE:To provide the receiver to be set to a proper reception state for each genre by muting a voice signal in response to a genre of a program. CONSTITUTION:The user presets a threshold value of muting by using a threshold level setting means 28, received data is separated into main frame data and subframe data by a data separation means 22, a sub synchronization detection protection means 23 detects the reception state of the subframe data, a genre identification means 25 extracts program genre information from the subframe data, whether or not muting is implemented is discriminated based on the program genre information extracted by a mute discrimination means 26, the set threshold level and the reception state, and when muting is to be implemented, a mute means 27 applies muting to the main frame data when the muting is to be implemented.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention automatically changes the level of a reception state to be muted (voice suppression) according to the genre of a program in a broadcast to which genre information of the program such as speech and music is added. The present invention relates to a receiving device having a function.

[0002]

2. Description of the Related Art A program received by a receiving device has various genre information such as speech and music, and a noise level that humans feel uncomfortable varies depending on the genre of the program. For example, in music and human speech, it is considered that speech can withstand listening even in a state where the speech is slightly noisier than music.

Among these receiving devices, DSR (Digita
l Satellite Radio) receiver is a multi-channel digital audio broadcasting using artificial satellites implemented in Germany, and is a CS (communication satellite) PCM that has been put to practical use in Japan.
It is a broadcast like an audio broadcast. In this multi-channel digital audio broadcasting, a digital audio program of 16 stereo (= 32 monaural) channels (sound quality equivalent to A mode of Japanese BS broadcasting) can be multiplexed for one repeater (transponder) of a satellite. Sampling frequency is 32 KHz, 16-bit data is compressed to 14-bit, BCH (Bose-Chaudhuri-Hocquenghem) error correction code, interleave, transmission scramble, differential encoding are applied, BCH error correction code, interleave, transmission scramble, difference Dynamic encoding is performed and QPSK modulation is performed for transmission.

In addition to the audio data that constitutes the main part of the main frame, information about a program called program information (hereinafter referred to as PA) is also transmitted in the special service frame which is a sub frame. The PA data includes a monaural / stereo distinction (2 bits) for programs of each channel, a program type number (4 bits) from “0” to “15” indicating the genre of the program as described below, and speech / music. It is data of discrimination (1 bit), mode (2 bits), and parity (1 bit). Among them, the distinction of speech / music is different from the distinction of the genre of the program, for example, distinguishing a speech part such as music introduction in a music program from an actual music broadcasting part (music part). It is also applied in the program.

The programs "0" to "15" are listed below, and "0" means no program type or undef.
ined, “1” is News (NEWS), “2” is Current Affair
s (AFFAIRS), "3" is Information (INFO),
“4” is Sport ((SPORT), “5” is Education (EDUC
ATE), "6" is Drama (DRAMA), "7" is Culture
(CULTURES), “8” is Science (SCIENCE),
"9" is Varied (VARIED), "10" is Pop Music
(POP M), “11” is Rock Music (ROCK M), “1
"2" is MOR Music (MOR M), "13" is Light Cl
assical (LIGHT M), "14" is Serious Classical
(CLASSICS), “15” is Other Music (OTHERM)
Is.

Now, the 16-stereo channel digital audio program multiplexing will be described in more detail.
As described above, voice is sampled at a sampling frequency of 32 KHz and quantized into 16 bits. A scale factor is introduced, a scale factor is assigned in units of 64 samples, and data is compressed from 16 bits to 14 bits. There is. This audio data is multiplexed in stereo with 16 channels (32 channels in the case of monaural) to form two bit strings of 10.24 Mbps, and QP
It is SK-modulated into one radio wave.

The transmitted data includes audio data converted into 14 bits, scale factor, program information, and program accompanying information. The data is 32K
It is framed in Hz, and the voice data contains 32 data of each channel for each frame, and the scale factor is such that the scale factor of each channel can be obtained for each special service frame composed of 64 frames. , 16 bits of data are contained in each frame, and the program accompanying information is assigned the surplus part of bits of this scale factor.

[0008] The program information is composed of one main so that the program category for all channels and the information on stereo / monaural may be entered for each special service superframe composed of eight special service frames (512 mainframes). Each frame contains 1-bit data. The frame is composed of two basic structures, an A frame and a B frame, having the same basic structure, and these A frame and B frame are simultaneously transmitted in parallel.

As shown in FIG. 2, the basic structure of each of the A frame and B frame is composed of 320 bits.
The content is composed of four 77-bit blocks BB, 11-bit Barker code BC and 1-bit special service bit SB. However, only the A frame has the special service bit SB, and the position of the B frame is fixed to "0".
Also, the B frame has an inverted Barker code.

As is apparent from FIG. 2, the A frame has 11 bits of the Barker code BC followed by 1 bit of the special service bit SB, followed by two 7 bits.
There are two 154 bits in which 7-bit bit blocks BB are alternately combined (interleaved) one by one, and the total is 320 bits. Barker code B for this
Transmission scrambling is applied to the portion excluding a total of 12 bits of C and the special service bit SB, further differentially encoded, and QPSK modulation is applied.

The structure of the 77-bit block is as shown in FIG. 3, and the stereo channel numbers are "1" and "2", "3" and "4", ... "15" and "16". Compose one 77-bit bit block BB. Explaining with a 77-bit block BB containing channel "1" and channel "2", first, the upper 11 bits of the left (L) audio data 14 bits of the channel "1" are arranged in order from the upper side, and then The upper 11 bits on the right (R) of the channel “1” are arranged in order from the upper one, and the upper 11 bits of the left 14 bits of the audio data of the channel “2” and the upper 11 bits on the right of the channel “1” are higher. Arrange in order.

These 44 bits are coded by BCH (6344), 19 check bits are added after, and additional information bits of channel "1" (bits containing the above scale factor or program accompanying information), One additional information bit of channel "2" follows each. After that, the lower 3 bits of the left audio data of the channel "1" are arranged in order from the higher order, and the lower 3 bits of the right of the channel "1", the lower 3 bits of the left of the channel "2", and the right lower of the channel "1". The lower 3 bits are each continued to make a total of 77 bits. The additional information bits and the lower bits are not error correction coded.

The special service bit SB constitutes a special service frame with 64 main frames (64 bits) and a special service super frame with 8 special service frames. The special service superframe consists of a program information special service superframe, a program source special service superframe, and a zero byte superframe (a special service superframe in which all bits except the Williams code are zero). There are three types of frames).

The program information special service superframe 7 frames, the zero byte superframe 1 frame, and the program source special service superframe 8 frames form a special service superframe. Special service frame consists of 16-bit William code and 2 bytes per stereo channel (1 byte per monaural channel)
6 bytes (48 bits) for stereo 2 channels, including the program information / program source byte of 1 and mode information byte of 1 byte per stereo channel. Since 16 stereo channels in total are multiplexed, the information of all channels can be transmitted in 8 frames of the special service frame, which is the special service super frame.

The information includes stereo channel numbers "1" and "2", "3" and "4", ..., "15" and "16", and channel numbers "1" and "2". In the special service frames other than the special service frame containing the information of "," the 11th and 12th bits of the Williams code are set to "1", which makes it possible to know the start of the special service superframe. ing. The special service superframe can recognize the break point by the existence of the zero-byte superframe.

As described above, the additional information bit is obtained from one mainframe for one bit per stereo channel, and is processed in units of 64 mainframes (64 bits). This additional information bit frame is synchronized with the above-mentioned special service frame, and the additional information bit of the frame next to the frame containing the last bit of the special service frame Williams code is the first of the additional information bit frame. Become a bit. The content of this additional information bit frame includes a scale factor of 3 bits for the left channel and a scale factor of 3 bits for the right channel (14,6) BCH.
It is 42 bits which is obtained by repeating 3 times the coded by, and 22 bits of program accompanying information.

This additional information bit frame is added to each stereo channel for every 64 main frames, that is, 16 types of additional information bit frames are obtained. Next, the DSR receiver will be described. FIG. 4 is a block diagram showing the configuration of this DSR receiver.
1 is a frequency conversion / tuning unit, and R is at its input end.
F input is input. The RF input is input from an antenna (not shown) or a CATV line. Frequency conversion /
By inputting the RF input to the tuning unit 1, the required frequency S1 is selected and sent to the QPSK demodulation unit 2.

The QPSK demodulator 2 demodulates the input signal S1 into two 10.24 Mbps digital signals S2a and S2b, which are an in-phase component and a quadrature component, and outputs them to the decoder 3. In the decoder 3, the two digital signals S2
a, S2b is received, the data is processed, and D / A
The voice data S3 is sent to the conversion unit 4 together with the necessary timing.
And the microcomputer 5 (hereinafter,
A signal S5 indicating program information, program accompanying information, etc. is output to a microcomputer. A channel selection signal and a stereo / monaural switching signal S4 are sent from the microcomputer 5 to the decoder 3. The switching signal S4 causes the decoder 3 to select and process necessary data.

FIG. 5 is a block diagram showing the internal structure of the decoder 3. In the synchronization detection protection section 3A of the decoder 3, the two digital signals S2a and S2b are used as the A frame data input and the B frame data input. In response to this, the differential decoding means 3A1 performs differential decoding and the synchronization detection means 3A2 detects the Barker code BC,
The synchronization protection means 3A3 establishes / protects frame synchronization. At this time, the special service bit SB immediately after the Barker code BC of the A frame is extracted and sent from the synchronization detection protection unit 3A to the bit processing unit 3B.

In addition to this, the synchronization detection protection unit 3A
Descrambling means 3A4 for descrambling transmission scrambling, deinterleaving means 3A5 for performing a part of the work of selecting the 77-bit block BB including the selected channel, and various timing pulses required for other blocks are generated. It has timing generation means 3A6. The 77-bit block BB selected by the deinterleaving means 3A5 is sent to the error correction section 3C.

In this error correction unit 3C, the BCH error correction means 3C1 ((63, 4) of the 77-bit block BB is used.
4) The BCH error is corrected, the corrected error is detected by the corrected error detecting means 3C2, and the result is sent to the bit synthesizing unit 3D. In the bit synthesizing unit 3D, the error-corrected upper 11 bits and lower 3 bits are combined by the bit synthesizing unit 3D1 to return to 14-bit data, and further, the special service frame majority / correction unit 3E2 of the data processing unit 3E described later. Using the scale factor obtained in step 3, the bit expansion means 3D2 restores 16-bit data.

This 16-bit data is the data interpolator 3
Sent to F. In the data interpolation unit 3F, the error correction unit 3
In F, the data interpolation means 3F1 performs a linear interpolation process on the error data detected by the error correction section 3C and determined together with the error information, and the D / A output means 3F2 outputs D to D in FIG. It is output to the / A converter 4.

Further, the special service bit processing means 3B1 of the bit processing portion 3B, which has received the special service bit SB from the differential decoding means 3A1 of the synchronization detection protection portion 3A, detects a Williams code, and the special service bit synchronization means 3B2. Establishes and protects the synchronization of the special service frame and the special service super frame by the timing generation means 3B.
3, the timing is changed so that the interface with the microcomputer 5 in FIG. 4 can be easily performed, and the data is output to the data processing unit 3E.

In the data processing unit 3E, the bit processing unit 3B
The PI processing means 3E1 processes the additional information bit of the channel received from the synchronization detection protection unit 3A and selected based on the synchronization information of the special service frame established in 1.
The scale factor was sent three times, and the special service frame majority decision /
A majority decision is made by the correction means 3E2, and further (14,
The BCH error correction of 6) is performed, and it is sent to the bit synthesizing unit 3D, and the program accompanying information is output together with the necessary timing signal.

Here, the LSI of the decoder 3 will be summarized with reference to FIG. The RF signal input to the DSR receiver is the QPSK demodulator 2 as described in FIG.
The data demodulated and reproduced by the terminal PDIN,
It is input to the differential decoder 11 through QDIN. This LSI is connected to a crystal oscillator (20.48 MHz) (not shown) at terminals XIN and XOUT, and this crystal oscillator and L
Clock recovery is performed with a PLL lock circuit (not shown) outside the SI.

The reproduced data input to the LSI is differentially decoded by the differential decoder 11 and then passed to the mainframe synchronization block 12. The mainframe synchronization block 12 detects the synchronization word and outputs two mainframes A (1
~ 8ch) and B (9 to 16ch) are identified, and when a frame including the selected channel, for example, 1ch is selected, the descrambler 13 performs the descrambling process on the main frame A, and the deinterleaver 14 deinterleaves the mainframe A. The processing is sequentially performed to separate the main frame A and the sub frame.

The separated sub-frame also has a sync word, and the special service frame sync block 15 detects sync. PA data representing a program genre is included in this special service frame. The subframe further includes an additional information ZI frame (see FIG. 3) having the same period as the special service frame, and a scale factor (as described above, an amplitude range used when expanding audio data from 14 bits to 16 bits). Value) and other data. This scale factor is selected by the scale factor majority / BCH error collector 16.

The main frame composed of sampled audio data and separated by the deinterleaver 14 is subjected to BCH decoding by the BCH error collector 17.
The data expansion unit 18 uses the scale factor 14
If the data decompressed from 16 bits to 16 bits and decompressed by the audio data interpolator 19 cannot be error-corrected by BCH decoding, it is interpolated, converted into audio data by the D / A (digital / audio) converter 20, and output. And the digital audio interface 2
At 0A, it is converted into a data format suitable for a digital audio interface and output.

[0029]

Such a receiving device is provided with a mute function for muting because the noise level varies depending on the genre of the program.
Mute is applied when the noise level reaches a predetermined level. However, even in the receiving device having these mute functions, one threshold value is set in the conventional DSR receiving device, and when the noise level exceeds the threshold value, the mute is applied. For this reason, there is a problem that the speech program is forcibly muted and the speech program cannot be listened to in a receiving state where the noise is anxious for the music but not so much for the speech.

The present invention has been made in view of the above-mentioned problems, and provides a receiving apparatus which can make the mute function more sophisticated by utilizing the genre information of a program and can mute according to the program genre. Especially.

[0031]

In order to achieve the above object, the present invention provides a data separation means for separating received data into mainframe data and subframe data, and a reception state of the subframe data. A reception state detection means for detecting, program genre information is extracted from the data of the sub-frame, a genre identification means for identifying the genre of the program, and a threshold value to which the user should apply sound suppression,
Whether or not to suppress the sound based on the setting means for setting according to the genre of the program, the identification result by the genre identifying means, the detection result by the receiving state detecting means, and the threshold value information set by the setting means. It is characterized in that a mute judging means for judging whether or not the sound and a sound suppressing means for suppressing sound on the data of the main frame based on the judgment result of the mute judging means are provided.

Further, according to the present invention, the receiving state detecting means is provided with a counter for indicating whether or not the data of the subframes are synchronized, and the receiving state is detected by the magnitude of the counter value. did. Further, the genre identifying means roughly divides the genre of the program into two categories, speech and music, and the genre setting means sets the threshold of the speech genre lower than the threshold of the music genre. And

In addition, according to the present invention, the reception state detecting means is provided with a counter for counting an error rate detected by error correction, and the reception state is detected by the magnitude of the error rate. Further, in the present invention, the genre identifying means subdivides the program genre into rock, pop music, classic, news, sports, etc., and the genre setting means includes the rock genre, pop music genre, classic genre, news. It is assumed that the threshold values of the genre, the sports genre, etc. are set with mutual priorities. further,
According to the present invention, the voice suppression unit forcibly suppresses the voice according to the priority of the genre setting unit, regardless of the reception state detected by the reception state detection unit.

[0034]

According to the present invention, the threshold value for the user to apply the voice suppression is set by the setting means according to the genre of the program, and the received data is made the main frame data and the sub frame data by the data separating means. The reception status of the separated subframe data is detected by the reception status detection means, and the program genre is identified from the subframe data by the genre identification means, and the identification result and the reception status detection means are detected. Based on the detection result and the threshold value set by the setting unit, the mute determination unit determines whether or not to apply the audio suppression, and when it is determined that the audio suppression should be applied, the main part of the reception data is detected. Apply audio suppression to frame data. As a result, the mute function can be made more sophisticated by using the genre information of the program, and muting according to the program genre can be performed.

[0035]

Embodiments of the present invention will be described below with reference to the drawings. In the description of this receiving device, the description will proceed with a DSR receiver. FIG. 1 is a block diagram of a receiving apparatus according to an embodiment of the present invention, showing a configuration of a portion after a differential decoder in a decoder of a DSR receiver.
In FIG. 1, the main frame output from the output terminal of the differential decoder is input to the main synchronization detection protection means 21 at the input terminal T3.

The term "synchronization protection" as used herein means to buffer the transition between the synchronous state and the asynchronous state.
This protection is divided into forward protection and backward protection, and backward protection is for detecting an incorrect sync word from an asynchronous state so as not to enter the pseudo sync state. When the regular sync word is garbled due to an error, it will not immediately go out of sync,
This is for maintaining the synchronization state.

The main frame detected by the main synchronization detection protection means 21 by the synchronization detection of the main frame is output to the data separation means 22. The data separating means 22 performs deinterleaving and descrambling processing to separate the mainframe from the mainframe and the subframe. The subframes separated by the data separation means 22 are input to the subframe synchronization detection protection means 23.

The sub-sync detection / protection means 23 inputs the sub-frame and synchronizes the special service frame. This part has a synchronization protection (forward protection) counter (64 stages) that takes a value of "0" in the asynchronous state and a value of "1" to "63" in the synchronization state. The larger the number, the more reliable the synchronization is. Therefore, the reception state can be monitored by the count value such that the reception state is good.

The main frame separated by the data separating means 22 is sent to the audio data processing means 24, where it is subjected to error correction and the like, and audio data is extracted and reproduced. On the other hand, the special service frame detected by the sub synchronization detection protection means 23 is sent to the genre identification means 25. The genre identifying means 25 extracts the program genre information in the special service frame, identifies whether the extracted program genre information is speech or music, and determines the mute determining means 26.
Send to.

Further, 28 is a threshold value setting means,
The threshold value setting means 28 is for setting a threshold value at which the user should apply audio suppression according to the program genre. The threshold value set by the threshold value setting means 28 is added to the mute judging means 26. Mute determination means 26
Is based on the threshold value set by the threshold value setting means 28, and based on the program genre information identified by the genre identifying means 25, the value of the synchronization protection counter of the sub-sync detection protection means 23 should be muted. It is determined whether it is smaller than the threshold value of the counter.

That is, among the program genres, speech has a smaller threshold than music. Therefore, speech is muted for the first time in a bad reception state than music. In other words, music is easier to mute. The mute judging means 26 judges whether or not the value of the sync protection counter is smaller than the threshold value of the sync protection counter to be muted. If the result of this determination is that the counter value is smaller than the threshold value, the mute means 27 mutes the audio data processed by the audio data processing means 24. That is, all bits of the audio data are set to "0".

Next, another embodiment of the present invention will be described. The main synchronization protection counter may be monitored as a method of changing the reception state monitor. Further, depending on the error correction code used, the reception state can be monitored even with the bit error rate detected at the time of error correction, so this value can also be used. That is, the threshold setting means 28
Thus, when setting the thresholds of speech and music, which are the major classifications of program genres, the threshold of speech is set higher than that of music.

Furthermore, instead of changing the mute method depending on the speech / music, for example, even for music, for listeners who do not want to hear rock,
The set is provided with a means for selecting the program genre, and in the case of a lock program, the genre is further improved by always muting regardless of the reception status (set the threshold to the maximum value that the counter can take). Mute is controlled by subdividing the mute and changing the mute judgment standard. In this case, it is possible for the listener to store information on whether or not to always mute each of the above 16 program types based on the setting.

Further, when the threshold value setting means 28 sets a threshold value for a small classification of programs such as rock, pop music, classical music, news, sports, etc., priority may be set. According to this priority order, the mute means 27 is operated by the sub sync detection protection means 23.
The mute may be forcibly applied regardless of the reception state of the data of the subframe detected in.

[0045]

As described above, according to the present invention,
Different thresholds for muting are set according to the genre of programs such as speech and music, and the reception status to be muted is changed according to the genre of the selected program. In addition, because only one threshold is set, noise is annoying for music, but in a receiving state where speech is not so noticeable, speech programs are also forced to mute and cannot be heard. Such inconvenience can be eliminated. Further, by setting the threshold value according to the program genre, it is possible to diversify the listening such that only the program of the desired genre is listened to and the programs of other genres are muted.

[Brief description of drawings]

FIG. 1 is a block diagram showing a configuration inside a decoder of a receiving apparatus according to an embodiment of the present invention.

FIG. 2 is a data configuration diagram of a 320-bit A frame applied to a conventional DSR receiver.

FIG. 3 is a data configuration diagram of a 77-bit block of the A frame of FIG.

FIG. 4 is a block diagram showing a configuration of a conventional DSR receiver.

5 is a block diagram showing an internal configuration of a decoder in the DSR receiving apparatus of FIG.

FIG. 6 is a decoder IC in a conventional DSR receiver.
It is a block diagram of.

[Explanation of symbols]

 21 Main Sync Detection Protection Means 22 Data Separation Means 23 Sub Sync Detection Protection Means 24 Audio Data Processing Means 25 Genre Identification Means 26 Mute Judging Means 27 Mute Means 28 Threshold Setting Means

Claims (6)

[Claims] 1. A data separation unit that separates received data into mainframe data and subframe data, a reception state detection unit that detects a reception state of the subframe data, and a program from the subframe data. Genre identification means for extracting genre information to identify the genre of the program, setting means for setting a threshold value for which the user should apply audio suppression according to the genre of the program, identification result in the genre identification means and the Based on the detection result of the reception state detecting means and the threshold value information set by the setting means, a mute judging means for judging whether or not voice suppression is applied, and based on the judgment result of the mute judging means, the main A receiving apparatus comprising: a voice suppression unit that applies voice suppression to frame data. 2. The reception according to claim 1, wherein the reception state detection means includes a counter for indicating whether or not the data of the subframes are synchronized, and the reception state is detected by the magnitude of the counter value. apparatus. 3. The genre identifying means roughly divides the program genres into two categories, speech and music, and the genre setting means sets the speech genre threshold value lower than the music genre threshold value. The receiving device according to claim 1. 4. The reception state detection means includes a counter that counts an error rate detected by error correction,
The reception state is detected based on the magnitude of the error rate.
The receiver described. 5. The genre identifying means sub-classifies program genres into rock, pop music, classic, news, sports, etc., and the genre setting means includes the rock genre, pop music genre, classic genre, news genre. The receiving device according to claim 1, wherein the respective threshold values of the sports genre, the sports genre, and the like are set with mutual priorities. 6. The receiving apparatus according to claim 5, wherein the voice suppressing unit forcibly suppresses voice according to the priority of the genre setting unit, regardless of the receiving state detected by the receiving state detecting unit.