JP3634296B2 - Digital broadcast receiver - Google Patents

Description

[0001]
BACKGROUND OF THE INVENTION
The present invention relates to a digital broadcast receiver, and more particularly, to an output mode when a received signal contains a certain amount of noise.
[0002]
[Prior art]
In digital broadcasting, audio information signals, image information signals, character information signals, and the like are compressed in a predetermined format, and information signals of a plurality of services are multiplexed and transmitted as data signals so as to be compatible with a plurality of services. For example, a DAB (Digital Audio Broadcasting: hereinafter abbreviated as DAB) system compliant with the European standard (Eureka 147) has been put to practical use as a system for transmitting and receiving data signals of the predetermined format including digital audio.
In the DAB system, one ensemble is composed of a plurality of services, and each service is also composed of a plurality of components. The details are described in the DAB standard, and therefore the description thereof is omitted.
[0003]
FIG. 6 is a diagram showing the structure of a transmission frame in the DAB system. The transmitted DAB signal is transmitted as an OFDM signal modulated by orthogonal frequency division multiplexing modulation, that is, OFDM (Orthogonal Frequency Division Multiplex), and the transmission frame (Transmission frame) shown in FIG. 6 is demodulated by demodulating the OFDM signal. Obtained from OFDM signal.
As shown in FIG. 6, the transmission frame in the demodulated OFDM signal is composed of a synchronization channel (Synchronization Channel), FIC (Fast Information Channel), and MSC (Main Service Channel) main block. The FIC transmits information necessary for the receiver to select a program and auxiliary information for the program, and the MSC transmits a subchannel of audio and data.
[0004]
The FIC for one frame is composed of three data blocks called FIB (Fast Information Block), and the MSC is composed of data blocks called CIF (Common Interleaved Frame) of 1 to 4 depending on the transmission mode. Time interleaving is applied. A plurality of subchannels are multiplexed in the CIF, and one subchannel corresponds to one program. Therefore, when the user intends to receive one program, the user selects and selects a specific subchannel from the CIF.
In a DAB system, if a certain ensemble is received, information on a plurality of services and components included in the ensemble can be obtained, and it is possible to instantaneously switch to a different service or component without changing the reception frequency. ing.
[0005]
A synchronization channel block is provided at the head of the transmission frame. This synchronization channel block includes a null signal portion for coarse synchronization and a reference phase symbol that bears a reference phase for differential QPSK demodulation in OFDM demodulation. Channel selection in the DAB receiver is performed in units of ensembles, and when other ensembles are received, the frequency is changed, that is, ensemble up / down is performed.
[0006]
FIG. 4 is a block diagram showing an example of a conventional digital broadcast receiving apparatus. Hereinafter, a conventional digital broadcast receiving apparatus will be described with reference to FIG. 4 using a DAB receiver as an example. This DAB will be described assuming that OFDM (Orthogonal Frequency Division Multiplex) is adopted as a modulation method.
In the digital broadcast receiving apparatus 50 shown in FIG. 4, the front end 13 receives a DAB carrier wave via the antenna 11 and processes the RF signal. In the signal processing, the RF signal is converted into an intermediate frequency signal (IF signal) that can be processed by a digital signal, and output to an analog / digital converter (A / D converter) 15. In the A / D converter 15, the analog signal is converted into a digital information signal and supplied to the I / Q detector 16. The I / Q detector 16, the FFT 17, and the DSP 18 constitute an OFDM demodulator 21.
[0007]
In the I / Q detector 16, the I signal and the Q signal are divided and supplied to the FFT 17 so that the OFDM signal can be processed in the subsequent stage. In the FFT 17, fast Fourier transform is performed using a DSP (digital arithmetic unit) 18 and supplied to the channel decoder 23. The channel decoder 23 performs de-interleaving for returning the order of signals to the original order, de-puncturing for returning the original data by inserting a code bit that has not been transmitted, and an error. Code detection and correction are performed. Further, the microprocessor (MPU) 31 calculates the code error rate together with the correction of the error code in cooperation with the channel decoder 23, and the code error rate is a predetermined value (this code error rate is described below). In this case, the mute control signal for stopping the decoding operation in the MPEG decoder 27 and the transmission of the decoded signal obtained by the decoding is changed from 0 to 1 (or from L to H). Changed and supplied to the D / A converter 29. The channel decoder 23 will be described in detail later.
[0008]
The compressed audio information is supplied from the channel decoder 23 to the MPEG decoder 27, where the compressed audio information is decoded and decompressed, and the digital / analog converter (D / A converter) 29 performs decoding. It is converted into analog information and output as an analog audio signal.
[0009]
FIG. 5 is a flowchart showing the procedure for generating the mute control signal in the conventional example. The flowchart shown in FIG. 5 shows processing performed by the channel decoder 23. In FIG. 5, first, in step S11, deinterleaving for canceling the frequency interleaving performed to randomize the error on the frequency axis and returning it to the original is performed, and the transmission frame shown in FIG. 6 is obtained.
[0010]
Next, the FIC and the MSC are separated in step S13. In step S15, Viterbi decoding is performed on the FIC data, and in step S17, a code error rate is calculated by an error rate calculation unit including the MPU 31 and the like. The steps S15 and S17 are actually performed almost simultaneously by the MPU 31 and the like. When the code error rate ERN exceeds a predetermined value (limit code error rate ERL), the MPEG audio signal is decoded and decoded. The mute control signal for stopping the transmission of the changed signal changes from 0 to 1 (or L to H) and is supplied to the D / A converter 29. The calculation of the code error rate ERN in step S17 is performed by comparing the data corrected by Viterbi decoding in step S15 with the data before error correction. Next, in step S19, the scramble is released (descrambled). In step S21, a CRC check is performed, and a mute control signal is supplied to the microprocessor (MPU) 31 that controls the entire digital broadcast receiving apparatus 50.
[0011]
On the other hand, in step S25, data of subchannels corresponding to the program selected by the user among the plurality of subchannels is separated. Next, in step S27, time deinterleaving for canceling time interleaving for randomizing errors on the time axis is performed. Incidentally, time de-interleaving is not performed for FIC. Then, Viterbi decoding is performed in step S29, descrambling is performed in step S31, and a DAB audio frame (DAB Audio frame) is output to the MPEG decoder 27 in FIG. In step S29, as in the case of FIC, detection and correction of code errors are performed on the convolutionally encoded data by Viterbi decoding.
[0012]
OFDM, which is a DAB modulation scheme, is a type of multicarrier modulation scheme, and is considered to be capable of stable reception that is less susceptible to multipath and fading in mobile communications. Nevertheless, the received data actually has some code error due to various causes. For this reason, as described above, the FIC or MSC data is error-corrected by Viterbi decoding, the error-corrected data is re-encoded, and the code error rate is calculated by comparing with the data before error correction. . When the code error rate exceeds a predetermined value, the operation of the MPEG decoder 27 is stopped and the output from the D / A converter 29 is stopped.
[0013]
[Problems to be solved by the invention]
However, the code error rate using Viterbi decoding does not always match the actual audible noise and distortion, and the code error rate is smaller than a predetermined limit code error rate (threshold). Nevertheless, noise and distortion may be conspicuous in the acoustic signal output from the speaker.
[0014]
On the other hand, depending on the receiving user and the type of broadcast source (signal source) to be received, there is a case where it is desired that the output is not interrupted even if there is a certain amount of noise or distortion. For example, if the received program is news or traffic information, the user wants to continue receiving if the broadcast content can be grasped even if there is a certain amount of noise or the sound quality is poor. Think.
In such a case, in the conventional digital broadcast receiving apparatus, the limit code error rate that is a threshold for starting the mute operation is fixed for each digital broadcast receiving apparatus and exceeds a preset limit code error rate. The output of the MPEG decoder 27 or D / A 29 is unconditionally stopped, and there is a problem that various requests generated for each user or for each case cannot be met.
[0015]
The present invention has been made in view of the above-described problems, and its object is to make it possible to change the mode of the mute operation as necessary, and to receive signals even when the received signal contains a relatively large amount of noise. Is to provide a digital broadcasting receiver.
[0016]
[Means for Solving the Problems]
The present invention has been made with the following configuration in order to achieve the above object.
According to a first aspect of the present invention, there is provided an error rate calculation means for calculating a code error rate of a received signal while receiving a digital broadcast, a code error rate of the received signal, a first limit code error rate, and a second code error rate. Output mode setting means for setting the output mode by comparing the magnitudes of the above, a decoding means for decoding received data of digital broadcast, a D / A converter for converting the output of the decoding means into an analog signal, and the D / A A digital broadcast receiving apparatus comprising noise reduction means for reducing noise of an analog audio signal output from the A converter, wherein the code error rate of the received signal is smaller than the first limit code error rate The analog audio signal is output substantially without going through the noise reduction means, and the received signal has a code error rate equal to or higher than the first limit code error rate and the second limit code error. When the code error rate of the received signal is larger than the second limit code error rate, the decoding means or the D / A conversion is output. The digital broadcast receiver is designed to stop the output of the receiver.
[0017]
According to a second aspect of the present invention, in the digital broadcast receiver according to the first aspect of the present invention, the digital broadcast receiver further includes an error rate setting means for setting the first limit code error rate or the second limit code error rate. Broadcast receiving device.
[0018]
A third invention is the digital broadcast receiving apparatus according to the first or second invention, wherein the noise reducing means is a band-pass filter that passes only signals in a predetermined frequency range. .
[0019]
A fourth invention is the digital broadcast receiving apparatus according to the second invention, wherein the first limit code error rate or the second limit code error rate can be set from the outside by an operator. .
[0020]
DETAILED DESCRIPTION OF THE INVENTION
The digital broadcast receiving apparatus of the present invention includes an output mode setting means for setting an output mode according to a code error rate of a received signal, a decoding means for decoding compressed reception data of the digital broadcast, and an output of the decoding means. A D / A converter for converting to an analog signal; and noise reduction means for reducing noise of the analog audio signal output from the D / A converter, wherein the code error rate of the received signal is a first limit code error When the rate is smaller than the rate, the analog audio signal is output substantially without going through the noise reduction means, and the code error rate of the received signal is not less than the first limit code error rate and not more than the second limit code error rate. When the analog audio signal is output via the noise reduction means, and the code error rate of the received signal is larger than the second limit code error rate It is obtained so as to stop the output of the decoding means or the D / A converter.
[0021]
Hereinafter, embodiments of the present invention will be described with reference to the drawings.
FIG. 1 is a block diagram showing an embodiment of a digital broadcast receiving apparatus of the present invention. 1, elements having the same functions as those in FIG. 4 are denoted by the same reference numerals, and the description thereof may be omitted in the following description.
In the digital broadcast receiving apparatus 10 shown in FIG. 1, it is assumed that the digital broadcast wave coming from the antenna 11 is the same as that of the conventional example explained in FIG. 4, and the audio information signal, image information signal, character information, etc. are predetermined. Compressed in this format and sent. In the digital broadcast receiving apparatus according to the embodiment of the present invention described below, a DAB (Digital Audio Broadcasting) receiver will be described as an example. In this DAB, description will be made assuming that OFDM (Orthogonal Frequency Division Multiplexing) is adopted as a modulation method.
[0022]
1 is different from FIG. 4 in that an output mode setting means 25, an error rate setting means 35, and a filter 33 are provided, and the frequency characteristic of the filter 33 depends on the output of the output mode setting means 25. This is a point controlled by the MPU 32. The filter 33 is, for example, a band-pass filter (band-pass filter) that passes only a signal in a predetermined frequency range, reduces noise included in the analog audio signal output from the D / A 29, and supplies the reduced noise to the next-stage audio amplifier. It is a filter for. The frequency characteristics of the filter can be switched by the MPU 32, and the filter can be substantially made through. The filter 33 may be a low pass filter.
[0023]
In the following description, a state where the filter 33 has a flat frequency characteristic is referred to as a filter OFF state or a through state, and a state where the filter 33 is a bandpass filter is referred to as a filter ON state. In the filter OFF state, the output of the D / A 29 is supplied to the next-stage audio amplifier without substantially passing through the filter 33, and in the filter ON state, the output of the D / A 29 is substantially passed through the filter 33. It is given to the next stage audio amplifier.
The output mode setting means 25 and the error rate setting means 35 are not configured by independent hardware, but their functions can be achieved by the microprocessor (MPU) 32 to which the input means is connected.
[0024]
In the channel decoder 23, as described in the conventional example, the code error of the code string in the demodulated OFDM signal output from the OFDM demodulator is detected using Viterbi decoding, and the error rate calculating means using the MPU 32 or the like Thus, the code error rate of the received signal is calculated. In the error rate setting means 35, the first and second limit code error rates are set or selected by the user and stored in the memory of the microprocessor (MPU) 32.
[0025]
In the output mode setting means 25, the code error rate (ERN) output from the channel decoder 23, the first limit code error rate (ERL) and the second limit code error rate (ERH) set or selected by the user. The output mode corresponding to the result is given from the output mode setting means 25 to the MPU 32. The MPU 32 controls the mute mode according to the output mode given from the output mode setting means 25. That is, ON / OFF of the filter 33 and execution / stop of decoding in the MPEG decoder 27 are controlled according to the output mode.
[0026]
Specifically, when the code error rate (ERN) of the received signal is smaller than the first limit code error rate (ERL), the filter 33 is substantially bypassed, that is, the filter 33 is made through. 33 outputs an analog audio signal to the audio amplifier, and the code error rate (ERN) of the received signal is not less than the first limit code error rate (ERL) and not more than the second limit code error rate (ERH). Sometimes, the filter 33 is turned on to output an analog audio signal. When the code error rate of the received signal is larger than the second limit code error rate (ERH), the decoding operation of the MPEG decoder 27 is stopped and the D / D The output of A29 is stopped.
When the decoded signal is not output from the D / A 29, power can be saved by not supplying power to the MPEG decoder 27.
[0027]
FIG. 2 is a flowchart showing a procedure for generating an output mode signal in the present invention. In FIG. 2, the same processing steps as those in FIG. 5 are denoted by the same reference numerals, and the description thereof may be omitted. 2 is different from FIG. 5 in that an output mode setting step S40 composed of steps S35 and S37 is provided in FIG. 2, and an output mode signal indicating the output mode is obtained in step S17. Between the first code error rate (ERN), the first limit error rate (limit code error rate) (ERL) and the second limit error rate (limit code error rate) (ERH) set or selected in step S35. Generated according to the comparison result. The code error rate ERN obtained in step S17 is a code error rate calculated using the FIC data being received, as described in the conventional example, and actually the FIC data separated in step S13. On the other hand, Viterbi decoding is performed, data that has been error-corrected by Viterbi decoding is re-encoded, error detection is performed in comparison with data before error correction, and a code error rate is calculated. The error rate calculation means for calculating the code error rate is actually composed of an MPU 32 or the like.
[0028]
In step S35, the user sets or selects the first limit code error rate ERL and the second limit code error rate ERH by the error rate setting means 35. When the user sets the limit code error rate, the user receives the digital broadcast with the filter 33 turned OFF, and when the reception state deteriorates until the user feels that the limit is acceptable, the user sets the error rate setting means 35. When the setting button is pressed, the code error rate ERN at that time is stored in the memory of the MPU 32 as the first limit code error rate ERL. In addition, when the reception state deteriorates until the user feels that the reception limit is reached when the filter 33 is turned on, the user presses the setting button of the error rate setting means 35 to change the code at that time. The error rate ERN is stored in the memory of the MPU 32 as the second limit code error rate ERH.
[0029]
When the user selects a limit code error rate, a plurality of options for the limit code error rate are preset in the digital broadcast receiving apparatus in advance for each of the first limit code error rate and the second limit code error rate. The plurality of prepared options are displayed on the display means 37, one of them is selected by the user, and the selected code error rate is set as the first or second limit code error rate to set the error rate. You may make it store in the memory of the means 35. FIG.
[0030]
In step S37, the code error rate ERN of the received signal is compared with the magnitudes of the first limit code error rate ERL and the second limit code error rate ERH, and the code error rate ERN of the received signal is compared with the first code error rate ERN. 00 is given to the MPU 32 as an output mode signal, and the code error rate ERN of the received signal is not less than the first limit code error rate ERL and not more than the second limit code error rate ERH. When 01, the output mode signal 01 is given to the MPU 32, and when the code error rate ERN of the received signal is larger than the second limit code error rate ERH, 11 is given as the output mode signal to the MPU 32.
[0031]
When the output mode signal is 00, the filter 33 is controlled to be OFF based on the control signal given from the MPU 32, and the output of the MPEG decoder 27 is not affected by the filter 33 and passes through the D / A 29. Is output. When the output mode signal is 01, the filter 33 is controlled to be ON based on the control signal given from the MPU 32, and the output of the MPEG decoder 27 is band-limited by the filter 33 to reduce noise. Is output. Further, when the output mode signal is 11, the decoding operation of the MPEG decoder 27 is stopped based on the control signal given from the MPU 32, or based on the control signal given from the MPU 32, or the D / A 29 The output is stopped and the mute operation is performed.
[0032]
On the other hand, from the MSC data separated in step S13, a specific subchannel is separated in step S25 in order to receive a desired program, and the separated data is subjected to time de-interleaving, Viterbi decoding, Scrambling is performed, and a DAB audio frame is output in step S31.
Note that the code error rate of the received signal obtained in step S29 may be adopted as the code error rate of the received signal used in step S37, instead of the code error rate of the received signal obtained in step S17. Alternatively, the output mode may be set using both the code error rate of the received signal obtained in step S17 and the code error rate of the received signal obtained in step S29.
[0033]
FIG. 3 is a flowchart showing an output mode setting procedure in the present invention. In FIG. 3, in step S51, the current code error rate ERN is calculated, and the process proceeds to step S53. In step S53, it is determined whether or not the current code error rate ERN is equal to or higher than the first limit code error rate ERL. If the result is affirmative, the process proceeds to step S55, and if not, the output mode signal [00] is output. Then, the process proceeds to step S57. In step S57, decoding by the MPEG decoder 27 is started or continued, and an analog audio signal is output from the filter 33 to the audio amplifier at the next stage while the filter 33 is OFF (through), and the process proceeds to step S63.
[0034]
In step S55, it is determined whether or not the current code error rate ERN is greater than the second limit code error rate ERH. If the result is affirmative, the output mode signal [11] is output and the process proceeds to step S59. If there is, the output mode signal [01] is output and the process proceeds to step S61. In step S59, the decoding in the MPEG decoder 27 is stopped or the output of the D / A 29 is stopped and muted, and the process proceeds to step S63. If power supply to the MPEG decoder 27 is stopped in step S59, the power consumption can be reduced. In step S61, decoding by the MPEG decoder 27 is started or continued, and an analog audio signal is output from the filter 33 to the audio amplifier in the next stage while the filter 33 is ON, and the process proceeds to step S63. In step S63, it is determined whether or not the reception cancel button has been pressed. If there is a cancel instruction, the process proceeds to step S65, and if there is no cancel instruction, the process proceeds to step S51. In step S65, the power supply of the tuner is turned off and this flow is stopped.
[0035]
As described above in detail, according to the embodiment to which the present invention is applied, the limit code error rate is not set to one fixed value at the time of manufacturing the digital broadcast receiving apparatus as in the prior art. The limit code error rate and the second code error rate are set. When the code error rate of the received signal is not less than the first limit code error rate and not more than the second code error rate, the audio signal that has been MPEG-decoded and converted into an analog signal is noise that uses a band-limiting filter. Noise is suppressed by the reducing means and output to the audio amplifier at the next stage.
[0036]
Therefore, when there is a lot of noise in the weak electric field area etc., it is possible to view broadcasts that could not be viewed due to the mute circuit operating, and continue to view important information such as news and traffic information. Is possible.
Further, by making it possible to set the first limit code error rate and the second limit code error rate according to the user's request, it becomes possible to respond to individual requests. In other words, it is possible to change the mode of the mute operation to an optimal mode by resetting the limit code error rate according to the user's personal feeling and the broadcast content. For example, in the case of music broadcasting, the critical code error rate is set low, and in the narration of news, traffic information, weather forecasts, etc., the critical code error rate is set large so that important news, traffic information, weather forecasts, etc. Reduces the risk of being missed.
[0037]
When a program type signal indicating whether the broadcast being transmitted is a video-centric broadcast, an audio-centric broadcast, an emergency broadcast, or the like is transmitted together with the broadcast content, the program type The limit code error rate may be automatically set based on the signal.
In the embodiment, when the code error rate of the received signal is smaller than the first limit code error rate, the analog audio signal is output substantially without the noise reduction unit. The analog audio signal is output through a first filter having a wide passband, and a code error rate of the received signal is greater than or equal to the first limit code error rate and less than or equal to the second limit code error rate. In some cases, even if the analog audio signal is output through a second filter having a narrower pass band than the first filter, it does not depart from the spirit of the present invention and the same effect can be obtained. Of course.
[Brief description of the drawings]
FIG. 1 is a block diagram showing an embodiment of a digital broadcast receiving apparatus of the present invention.
FIG. 2 is a flowchart showing a procedure for generating an output mode signal in the present invention.
FIG. 3 is a flowchart showing an output mode setting procedure in the present invention.
FIG. 4 is a block diagram showing an example of a conventional digital broadcast receiving apparatus.
FIG. 5 is a flowchart showing a procedure for generating an output mode signal in a conventional example.
FIG. 6 is a diagram showing a configuration of a transmission frame in the DAB system.
[Explanation of symbols]
21 OFDM demodulator
23 channel decoder
25 Output mode setting means
27 MPEG decoder
29 D / A converter
32 Microprocessor (MPU)
33 Noise reduction means (filter)
35 Error rate setting means
37 Display means

Claims (4)

An error rate calculating means for calculating a code error rate of FIC data included in a received signal during reception of a digital broadcast by a DAB system ;
Output mode setting means for setting the output mode by comparing the code error rate of the received signal with the first limit code error rate and the second code error rate;
Decoding means for decoding received data of digital broadcasting;
A D / A converter for converting the output of the decoding means into an analog signal;
Bei example a noise reduction means for reducing the noise of the analog audio signal outputted from the D / A converter,
When the code error rate of the received signal is smaller than the first limit code error rate, the analog audio signal is output substantially without going through the noise reduction means,
When the code error rate of the received signal is not less than the first limit code error rate and not more than the second limit code error rate, an analog audio signal is output through the noise reduction means,
The digital broadcast receiving apparatus, wherein when the code error rate of the received signal is larger than the second limit code error rate, the output of the decoding means or the D / A converter is stopped. 2. The digital broadcast receiving apparatus according to claim 1, further comprising error rate setting means for setting the first limit code error rate or the second limit code error rate. 3. The digital broadcast receiving apparatus according to claim 1, wherein the noise reduction means is a band-pass filter that passes only a signal in a predetermined frequency range. 3. The digital broadcast receiving apparatus according to claim 2, wherein the first limit code error rate or the second limit code error rate can be set from the outside.

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