JPH08331067A - Voice signal processing unit for bs tuner - Google Patents

Abstract

PURPOSE: To extend a voice output time even when a radio wave state is deteriorated and to avoid production of noise sound when voice mute is released. CONSTITUTION: When a state of a radio wave is deteriorated, a range bit replacement section of a voice noise reduction circuit 61 replaces range bits with prescribed range data generated internally and when out of frame synchronism is detected, a synchronizing signal replacement section replaces synchronizing signal bits with a synchronizing signal generated internally. When the state of the radio wave is furthermore deteriorated and a change in a control code such as change in A mode/B mode is detected, a voice mute control section 71 outputs a voice mute signal to mute a voice signal. On the other hand, the state of the radio wave is improved and a synchronizing signal is detected continuously for a prescribed number of times or over, the voice mute control section 71 releases the voice muting.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an audio signal processing device for a BS tuner, and in particular, even if the condition of the satellite broadcast radio wave is deteriorated, noise-reduced audio is output as long as possible and the radio wave condition is deteriorated. The present invention relates to an audio signal processing device of a BS tuner capable of preventing abnormal noise from occurring when the audio mute is released after the audio mute is applied.

[0002]

2. Description of the Related Art In satellite broadcasting, a transmitting station uses a synchronization signal, a control code, range data, data-compressed M channels, and N sampling audio data for each frame (M = 4, N = 32 as described later). Or M = 2, N = 4
8) is included in the audio signal, and the audio signal is 5.727272MH.
DPSK modulation (Differential Phase Shift Key)
ing: differential phase modulation), the DPSK modulated signal is multiplexed with a 4.5 MHz video signal, and the 14 GHz main carrier wave is FM-modulated by the multiplexed signal and transmitted to a broadcasting satellite. The broadcasting satellite frequency-converts the radio wave from the transmitting station, amplifies it, and radiates it again toward the ground. The satellite broadcast receiver receives the broadcast radio waves sent from the broadcast satellite, separates the video signal and the audio signal after FM detection, and then performs a predetermined process to convert the video signal and the audio signal into the video device and the audio device. Output to.

FIG. 6 is a frame configuration diagram of a voice signal.
One frame consists of 2048 bits and 1000 times per second
(The frame frequency is 1 KHz, and the cycle is 1 ms). The first 16-bit code in the frame structure indicates the beginning of the frame and is a synchronization signal for facilitating the reproduction of the bit clock on the receiving side.
It has a bit pattern of "001001101011110". The next 16-bit code of the frame sync signal is a control code (control signal). As shown in FIG. 7, the A / B mode is distinguished, the television sound and the independent sound are distinguished, the stereo and monaural are distinguished, and the mode is switched. This is for distinguishing whether or not there is control for suppressing unnecessary noise that occurs when switching between program transmission stations and the like.

In the A mode, an analog audio signal for one channel is quantized into 14-bit data at a sampling frequency of 32 KHz, and then quasi-instantaneously compressed to 10-bit data (described later), and four channels are transmitted in one frame. The transmission speed is 32 KHz × 10 (bits) × 4 (channels) = 1280 Kb / s. Since 1 frame is 1ms, 1 in A mode
32 quantized data per channel (PC
M voice data) is included. In the B mode, an analog audio signal for one channel is quantized into 16-bit data at a sampling frequency of 48 KHz, and two channels are transmitted in one frame. The transmission rate is 48 KHz x 16 (bits) x 2 (channels) = 1536 Kb / s, and one frame contains 48 quantized data (PCM audio data) per channel. Since independent data and codes for error correction as well as voice data are transmitted, the code transmission rate including these codes is 2048 in both A and B modes.
Kb / s. FIG. 8 is a table for explaining the specifications of the voice coding method in the A and B modes.

Returning to FIG. 6, the next 32 bits (A mode) and 16 bits (B mode) of the control code are range bits, and as shown in FIG. 9, 8 bits are assigned to each channel. There is. Of the 8 bits, the first 3 bits are the range bits corresponding to the range number, the next 4 bits are the bits for correcting the range bit error that occurred during transmission, and the remaining 1 bit is empty.

As described above, in the A mode, quantization is performed with 14 bits, and quasi-instantaneous compression is performed with 10 bits for transmission. In this quasi-instantaneous compression method, as shown in FIG. 10 (A), it is determined in which range, 0 to range 4, the 14-bit audio signal level exists, and as shown in FIG. 10 (B),
It is a method of compressing 14-bit data into 10-bit data by (1) the range, (2) upper 9 bits of 14-bit audio data, and (3) 1-bit code. That is, 1
The maximum level of the audio signal is detected for each channel, it is determined which range of the range 0 to range 4 the maximum level exists, the existing range is expressed by 3 bits, and the audio signal level of the channel is determined. Expressed in 10 bits. In the B mode, the sound is 2 channels, so
The range bits are 16 bits which is half of the A mode, and the range bit portions corresponding to channels 3 and 4 of the A mode are used as independent data.

After the range bit, in the A mode, four channels of audio data are arranged for 32 samplings per channel, and then independent data and error correction code are arranged to form one frame. In the B mode, audio data for two channels is arranged for 48 samplings per channel, and then independent data and error correction code are arranged to form one frame. The last 7 × 32 bit error correction code of the frame has a high error correction effect and a BCH (6
3,56) code is adopted. In this code, 56 bits of the bit length 63 are information, and the remaining 7 bits are used for error correction. Among 56 bits, 1 bit error that occurred in the transmission path can be corrected and 2 bit error can be detected. It is like this.

The codes (2048
(Bit) is rearranged in units of 32 bits when transmitted (interleave). FIG. 11 is an interleaved matrix showing the sorting rules. The upper row is for A mode, the lower row is for B mode, and 32 ×
It is composed of 64 (= 2048) squares. Each frame of 2048-bit data is written in the memory in order in the horizontal direction, and one subframe is composed of 64 bits in the horizontal direction, has 32 subframes in total, and is read out from the memory in order in the vertical direction (arrow direction). Send. A subframe consists of one sync signal bit or control signal bit, one range bit,
Each is composed of 10 4-channel audio data, 15 independent data, and 7 error correction codes.
The 7-bit BCH (63,56) code is added to 56-bit information data excluding the sync signal bits or control signal bits to correct errors in these 56-bit information data. Errors in signals and control signals cannot be corrected. The reason for interleaving is
This is because even if a code is continuously lost or erroneous during transmission, the error is dispersed when the code is rearranged in the original order on the receiving side, and error correction (interpolation) is facilitated.

The code transmitted by bit interleaving may be consecutive 0 or 1 depending on the signal content. Even in such a case, a pseudo random signal is added so that 0 or 1 appears irregularly in order to facilitate the reproduction of the bit clock on the receiving side (scramble processing). The receiving side can descramble and restore using the same pseudo-random signal as the transmitting side. Then 5.72 with the scrambled digital signal
The 7272MHz subcarrier is DPSK-modulated, and the DPSK-modulated signal is adjusted to 4.5.
It is multiplexed with a video signal of MHz, and the 14 GHz main carrier is FM-modulated from the multiplexed signal and transmitted to a broadcasting satellite.

FIG. 12 is a block diagram of the transmitting side. Reference numeral 1 is a sampling circuit, which quantizes four channels of analog audio signals in the A mode, and quantizes two channels of analog audio signals in the B mode, and 2 represents audio data by a quasi-instantaneous compression method. 6 is a coding circuit for compressing audio data of each channel to form a frame shown in FIG. 6 and outputs the frame, 4 is an interleave circuit for performing a bit interleaving process, and 5 is a pseudo random signal. Scramble circuit for performing scramble processing by 6
Is a 4-phase DPSK circuit that DPSK-modulates the 5.727272MHz subcarrier with a scrambled digital signal, and divides the digital signal into groups of 2 bits each, (0,0), (1,0), (1,1 ),
The phase of the subcarrier is changed by associating (0, 1) with 0 ⁰ , 90 ⁰ , 180 ⁰ , -90 ⁰ . Reference numeral 7 is a synthesizing section for multiplexing a DPSK modulated signal with a 4.5 MHz video signal, and 8 is an FM modulating section for FM-modulating a 14 GHz main carrier from the multiplexed signal.

FIG. 13 is a block diagram of a satellite broadcast receiving apparatus, which is mounted on a vehicle. 11 is a BS antenna, 12 is a 14 GHz band satellite broadcast signal captured by the BS antenna at 1 GHz
A BS converter that converts to a band BS-IF signal (first frequency conversion) and then amplifies to a signal level required by the BS tuner. 13 extracts a video signal and an audio signal from the BS converter output signal. A BS tuner that controls a direction of a BS antenna while outputting after performing predetermined video processing and audio processing, 15 is a video device such as a television, 1
Reference numeral 6 is an audio circuit, 17 is a speaker, and 18 is a motor for changing the direction of the BS antenna, for example, a stepping motor.

In the BS tuner 13, 13a is a frequency conversion section (second frequency conversion section) for mixing a local oscillation signal determined by tuning and a BS-IF signal and converting it into a second intermediate frequency signal, and 13b is an intermediate frequency conversion section. An intermediate frequency amplification unit that amplifies the frequency signal, 13c performs FM detection, a video signal is separated by a 4.5MHz low pass filter, and an audio signal is separated by a 5.73MHz band pass filter. 13d is a video signal. A video signal processing unit that outputs a video signal by performing a predetermined process on the audio signal processing unit 13e, an audio signal processing unit 13e that performs a predetermined process on the audio signal and outputs an audio signal
Is a CN ratio detector for detecting a CN ratio (Carrier to Noise ratio). The CN ratio is the ratio of the magnitudes of the carrier signal and the noise signal contained therein, and indicates the radio wave condition. Reference numeral 13g is an antenna control unit for controlling the BS antenna 11 so as to face the broadcasting satellite, 13g-1 is a microcomputer, and 13g-2.
Is a gyro that detects the traveling direction of the vehicle, and 13g-3 is a motor controller. When the CN ratio falls below the threshold level, the microcomputer 13g-1 rotates the BS antenna 11 and directs the BS antenna in the direction in which the CN ratio becomes the largest.
Also, the microcomputer 13g-1 will change when the vehicle changes direction.
A change in the traveling direction is obtained from the gyro 13g-2, and the BS antenna 11 is rotated in the direction opposite to the changing direction so that the antenna is controlled to face the satellite.

FIG. 14 is a block diagram of the audio processing unit 13e, in which 21 is a 5.73 MHz band pass filter for extracting the audio PCM subcarrier, and 22 is a bitstream signal which is a data string by phase-detecting the audio PCM subcarrier. Demodulate
A QPSK demodulator, which detects the phase of an input modulated signal with two quadrature carriers synchronized in phase with the modulated signal, converts the signal obtained by phase detection into a digital signal with a data strobe circuit, and then outputs the differential signal. The decoding circuit operates to restore the original signal. Reference numeral 23 is a PCM demodulation unit, 24 is a digital filter, and 25 is a DA for converting a digital audio signal into an analog signal.
It is a converter. The PCM demodulation unit 23 is the QPSK demodulation unit 22.
The descrambling circuit 23 receives the data stream signal from the descrambling circuit 23 and performs descrambling, deinterleaving, error correction, data decompression, and the like on the contrary to the transmitting side.
a, deinterleave circuit 23b, error detection and correction circuit 2
3c, companding decoding unit 23d, range bit extraction error correction circuit 23e, synchronization signal detection protection circuit, control signal detection unit 23
g, a signal separation circuit 23h for separating an audio signal, and an error correction circuit 23i for interpolating and outputting the preceding and subsequent audio data without using the audio data when an error of 2 bits or more is detected in the audio data. ing.

The sync signal detection protection section 23f detects the sync signal and protects it by using the correctly detected sync signal when the sync signal cannot be detected. The signal processing standard is a synchronization signal indicating the beginning of a frame, which is transmitted for each frame. If this frame synchronization signal cannot be detected correctly, all operations cannot be performed accurately.
Therefore, when it cannot be correctly detected due to noise or the like, the correctly detected synchronizing signal is used to generate a synchronizing signal to prevent malfunction of the synchronizing signal that has not been detected or lost. The control signal detector 23g detects a 16-bit control code. Since the control signal is sent frame by frame, the control signal detector 23g makes a majority decision using the control signal of n (for example, 36) frames to identify the control signal and protect it against an error. For example, if A mode is detected 19 times or more among 36 frames, mode A
To determine. The control signal is used to detect the mode, the state of the sound being transmitted, the presence or absence of scramble, and the like, which is used for the sound mode display, the image, and the sound mute. Also, A / B
The mode signal is input to the DA converter 25 to execute the DA conversion processing in synchronization with the sampling frequency according to the mode.

[0015]

In the BS tuner, when the radio wave is weak, a harsh voice noise is generated. This noise is mainly caused by an error in the range bit and a loss of the sync signal. Therefore, the following controls have been proposed. That is, the range bits are changed so that the range becomes 3 or 4 (range with low audio level) when the radio wave weakens and the CN ratio deteriorates.
When the CN ratio is good, the position of the synchronization signal is stored and C
When the N ratio deteriorates, the sync signal bits are replaced with the correct sync signal pattern. When the CN ratio further deteriorates, the audio mute is applied, when the CN ratio improves, the audio mute is released, and when the radio wave condition improves and the CN ratio becomes good, the replacement of the range bit and the synchronization signal bit is stopped. ing. According to this method, CN
Even if the ratio deteriorates, unpleasant voice noise can be reduced. It is the A mode that noise can be reduced by changing the range bit.

By the way, in the proposed method, CN
Focusing on the fact that the number of code error detections increases when the ratio deteriorates, switching start, audio mute start, audio mute release, and switching stop are performed based on the number of error detections. However, in the above method, since the sound is muted based on the number of times of error detection, there is a problem that the sound is muted even when the sound with reduced noise can be output. Such a problem becomes a particularly serious problem in a vehicle-mounted satellite broadcast receiver. This is because the receiving environment of the vehicle-mounted satellite broadcast receiver is constantly changing because the vehicle frequently changes its traveling direction and passes through areas with different reception conditions. That is, when the above method is applied to such a vehicle-mounted satellite broadcast receiver, even when noise-reduced audio can be output, audio mute is applied early and the time during which audio is output is shortened, Moreover, audio mute is frequently applied.

Further, in the above method, since the audio mute is canceled based on the number of times of error detection, there is a problem that an abnormal noise is generated when canceling the audio mute. This is because the control code has not yet been determined when the audio mute is released. That is, the control signal is n frames (n
Is determined by the majority decision of (36), for example, and is not determined until 36 frames (36 ms) have elapsed after the audio mute is released. Therefore, during that time, the wrong mode is specified to the DA converter, and the DA conversion cannot be performed correctly.
Also, when the audio mute is released, if the mode is erroneously recognized, the DA converter cannot be correctly designated and an abnormal noise occurs.

From the above, the first object of the present invention is to extend the timing of muting audio as much as possible even when the radio wave condition deteriorates, which makes it possible to prolong the time during which audio is output, and An object of the present invention is to provide an audio signal processing device of a BS tuner that can prevent a situation where audio mute is frequently applied. A second object of the present invention is to provide an audio signal processing device for a BS tuner that does not produce an abnormal noise when the audio mute is released.

[0019]

According to the present invention, the first object is to provide a voice signal including a sync signal, a control code, range data, data-compressed M channel, and N sampled voice data for each frame. A BS that demodulates a signal, performs error detection and correction processing on the demodulated bit stream and outputs it, and inputs audio data obtained by performing expansion processing on the audio data in the error-corrected bit stream to a DA converter. An audio signal processing device of a tuner, a radio wave condition monitoring unit for detecting a frame sync signal from a bit stream, and a radio wave condition monitoring for judging deterioration of a radio wave condition based on an error detection frequency detected by an error detection and correction process. Section, a frame synchronization signal generation section that internally generates a frame synchronization signal, and a predetermined range data that is internally generated A range data generator, a replacement timing generator that generates a synchronization signal bit replacement timing and a range bit replacement timing in the bitstream, and a range bit in the bitstream when the radio wave condition deteriorates. A range bit replacement unit that replaces the range data output from the generation unit at the range bit replacement timing, and when a frame synchronization loss is detected, synchronizes the synchronization signal bit in the bit stream with the synchronization signal output from the synchronization signal generation unit. When the synchronization signal replacement unit that replaces at the signal replacement timing, the control code monitoring unit that detects an error of a predetermined bit in the control code when the radio wave condition deteriorates, and the error of the predetermined bit is detected by the control code monitoring unit , Audio mute And the audio mute control unit for outputting a No. is achieved by the BS tuner of the audio signal processing apparatus having an audio muting circuit to mute the sound based on the audio mute signal. Further, according to the present invention, the second object is achieved by the audio signal processing device of the BS tuner that cancels the audio mute when the audio mute control unit continuously detects the synchronization signal a predetermined number of times or more.

[0020]

According to the present invention, the audio signal including the synchronizing signal, the control code, the range data, the data-compressed M channel, and the audio data for N samplings is demodulated for each frame, and the demodulated bit stream is subjected to error detection and correction processing and output. Then
In a voice signal processing device of a BS tuner for inputting voice data obtained by subjecting voice data in the error-corrected bit stream to expansion processing,
When the radio wave condition deteriorates, the range bit replacement unit replaces the range bit with the predetermined range data generated internally, and the synchronization signal replacement unit internally generates the synchronization signal bit when the loss of frame synchronization is detected. Replace with the sync signal. Then, when the radio wave condition further deteriorates and an error of a predetermined bit (for example, a mode bit of A mode / B mode) in the control code is detected, the audio mute control unit outputs an audio mute signal to mute the audio. On the other hand, when the radio wave condition is improved and the synchronization signal is continuously detected a predetermined number of times or more, the audio mute control unit cancels the audio mute.

As described above, since the error of a predetermined bit in the control code (for example, the mode bit of the A mode / B mode) is detected and the audio mute is performed, the radio wave condition deteriorates and the A mode / B is deteriorated. Audio can be output until the mode changes, and the timing for audio mute can be extended as much as possible, which can extend the time during which audio is output and prevent frequent audio mute situations. it can. As long as the mode identification is correct, the DA converter can be instructed to the correct mode, the DA converter can perform the DA conversion based on the correct sampling frequency according to the mode, and the voice can be output. In addition, the audio mute is canceled when a sync signal is detected more than a predetermined number of times in succession, for example, the number of frames used for majority decision, so the mode is definitely specified when the audio mute is canceled,
It is possible to prevent generation of abnormal noise due to mode indetermination.

[0022]

【Example】

(a) Audio signal processing device for BS tuner FIG. 1 is a block diagram of an audio signal processing device for a BS tuner according to the present invention. Reference numeral 51 is an audio signal processing unit, which extracts an audio PCM subcarrier from the BS-IF signal output from the BS converter and performs QPSK demodulation, descrambling, deinterleaving, BCH error detection and correction processing, range bit correction processing, and audio data. Expansion (restoration) processing, mute control,
What executes DA conversion processing, etc. 61 is an audio noise reduction circuit that reduces the audio noise by replacing the range bit and sync signal bit when the radio wave condition deteriorates, and 71 deteriorates the radio wave condition so that the sound cannot be output correctly. When I did
It is an audio mute control unit for applying audio mute and canceling the audio mute when the radio wave condition improves.

In the audio signal processing unit 51, 51a is a 5.73 MHz bandpass filter for extracting an audio PCM subcarrier, and 51b is a QPSK demodulation unit for phase-detecting the audio PCM subcarrier and demodulating a bit stream signal which is a data string. , 51c receives a data stream signal from the QPSK demodulation unit 51b, and performs a descrambling circuit which performs a descrambling process in reverse to the transmitting side, 51d represents a deinterleave circuit,
51e is a BCH error detection / correction circuit, 51f is a companding decoding unit for expanding 10-bit audio data to 14 bits, 51g is an audio mute circuit, 51h is a DA converter, 51i is a BCH correction circuit for range bits, and 51j is synchronous. A sync signal detecting unit for detecting a signal, and 51k is a control code detecting unit. On the output side of the companding decoding unit 51g, there are provided a signal separation circuit for separating an audio signal, an error correction circuit for interpolating the preceding and following audio data when an error of 2 bits or more is detected in the audio data, a digital filter, etc. Exists but omitted.

(B) Audio Noise Reduction Circuit / Configuration FIG. 2 is a configuration diagram of the audio noise reduction circuit. 61a is Q
A synchronization detection circuit that detects a frame synchronization signal from the bit stream output from the PSK demodulation unit 51b, 61b is a 2048-bit counter that counts the clock signal reproduced from the received signal, and the detection timing of the frame synchronization signal when the radio wave condition is good Count the clock in synchronization with
A bit position in the frame of the received bit is output, 61c is an error flag comparator for identifying an error flag output from the BCH error detection / correction circuit 51e when an error is detected, and 61d is detected during one frame (1 ms). It is an error flag counter that counts the number of error flags. The synchronization detection circuit 61a outputs the signal SYO when the frame synchronization signal cannot be detected at the regular position, that is, when the loss of synchronization is detected, and stops the output of the signal SYO when the synchronization signal is detected. Also,
The error flag counter 61d outputs a signal EWB when the number of error flags becomes equal to or larger than the first set value TH1 and the deterioration of the radio wave state (deterioration of the CN ratio) is detected. Second set value TH2 (<T
When it becomes H1), the output of the signal EWB is stopped.

61e indicates that when deterioration of the CN ratio is detected,
An operation control unit for instructing replacement of range bits and for instructing replacement of synchronization signal pattern when out-of-sync is detected, 61f is a frame synchronization signal generation circuit for internally generating a frame synchronization signal pattern "0001001101011110", 61g Is range data “11010010” or “0” indicating the range with lower audio level (eg range 3 or 4).
This is a range data generator that internally generates data obtained by multiplying "0111010" by a scrambled bit string. 61h
Is a first switch, which normally allows the bit stream output from the BCH error detection / correction circuit 51e to pass therethrough, but the synchronization signal output from the frame synchronization signal generation circuit 61f only when replacement of the synchronization signal is instructed. The pattern switch 61i is a second switch, which normally allows the bit stream output from the first switch 61h to pass therethrough, but only when range bit replacement is instructed. It passes the range data output from.

Reference numeral 61j is a replacement timing generation circuit for generating the replacement timing of the range bit and the synchronization signal bit. Replacement timing generation circuit 61j
When the operation control circuit 61e instructs the range bit replacement, the range bit position is identified from the bit position output from the bit counter 61b, and the range bit replacement is performed by the range data generation circuit 61g and the second switch. Instruct 61i. Further, the replacement timing generation circuit 61j receives the bit counter 61b when the operation control circuit 61e instructs the replacement of the synchronization signal bit.
The synchronization signal bit position is identified from the bit position output from the frame synchronization signal generation circuit 61f and the first switch 61h is instructed to replace the synchronization signal bit.

When the radio wave condition is good, the number of error flags is small, and the synchronization is not lost, the bit stream is descrambled and deinterleaved via the first and second switchers 61h and 61i. After the error detection and correction, the companding decoding unit 51f and the BCH correction circuit 5 for range bits
Input to 1i. The BCH correction circuit 51i corrects an error in the range bit and notifies the companding decoding unit 51f of the range. The companding decoding unit 51f determines 1 based on the notified range.
The 0-bit audio data is expanded into 14-bit audio data and output to the DA converter 51h. DA converter 5
Reference numeral 1 converts the voice data into an analog voice signal and inputs it to an external audio circuit.

The radio wave condition deteriorates and the number of error flags is first.
Error flag counter 61 when the set value TH1 or more is exceeded.
The signal EWB is generated at d. When this signal is generated, the operation control circuit 61e causes the replacement timing generation circuit 61j.
Instruct to replace the range bit. As a result, the replacement timing generation circuit 61j causes the bit counter 61
The range bit position is identified from the bit position output from b and the range data generation circuit 61g and the second switch 61i are instructed to replace the range bit. The range data generation circuit 61g immediately generates range data,
The switch 61i of (1) passes the range data, and thereafter passes the bit stream. The operation control circuit 61e instructs replacement of the range bits until the signal EWB is no longer output from the error flag counter 61d, and replaces the range bits with the predetermined range data generated internally as described above. As a result, the signal condition deteriorates, and even if the range is erroneous, a large noise is not output because the range is changed to a low level range.

When the radio wave condition deteriorates and out-of-sync is detected, the sync signal detection circuit 61a generates the signal SYO. When this signal is generated, the operation control circuit 61e instructs the replacement timing generation circuit 61j to replace the synchronization signal bit. As a result, the replacement timing generation circuit 61j identifies the synchronization signal bit position from the bit position output from the bit counter 61b and instructs the frame synchronization signal generation circuit 61f and the first switch 61h to replace the synchronization signal. Frame synchronization signal generation circuit 6
1f immediately generates a sync signal pattern, and the first switch 61h allows the sync signal pattern to pass, and thereafter allows a bit stream to pass. The operation control circuit 61e waits until the signal SYO is no longer output from the synchronization detection circuit 61a.
The replacement of the synchronization signal is instructed, and the synchronization signal bit is replaced with the internally generated synchronization signal as described above. As a result, even if the signal condition deteriorates and synchronization loss (including non-detection of synchronization signal) occurs, the synchronization signal pattern can be relocated to the correct position, signal processing can be performed accurately, and noise generation can be prevented. You can

(C) Audio Mute Control Unit-Configuration FIG. 3 is a block diagram of the audio mute control unit. Reference numeral 71a is a control code for monitoring a change in a predetermined bit in the control code (control signal), for example, an A / B mode identification bit, detecting an error and outputting a control code change detection signal CCD when the radio wave condition deteriorates. Monitoring circuit, 71b is a synchronization detection signal SDT
Is a synchronization detection circuit for monitoring whether or not a synchronization signal is detected, and 71c is an audio mute control circuit. The audio mute control circuit 71c outputs an audio mute signal SMUT and a signal NLSP instructing to stop the operation of the audio noise reduction circuit 61 when a change in the control code is detected in the state where the sync signal is not detected. Then
When the sync signal is continuously detected a predetermined number of times or more, for example, the number of frames used for the majority decision of the control code (= 36) or more, the audio mute is released and the reduction operation of the audio noise reduction circuit 61 is permitted. The audio mute control circuit 71c, when the audio suppression is instructed,
Also, an audio mute signal is output when the channel is selected.

When the radio wave condition deteriorates and the sync signal cannot be detected, the A / B mode change indicates that the control code monitoring circuit 71a.
When detected by, the audio mute control circuit 71c outputs the audio mute signal SMUT and the operation stop signal NLSP. When the audio mute signal SMUT is input, the mute circuit 51g blocks the audio data output from the companding decoder 51f, and blocks the audio data from inputting to the DA converter 51h. In addition, the operation control circuit 61e of the audio noise reduction circuit 61 uses the operation stop signal NLSP.
Stops the range bit and sync signal bit replacement operation.

After that, when the radio wave condition is improved and the synchronization signal is continuously detected a predetermined number of times (= 36 times) or more, the audio mute control circuit 71c stops the output of the audio mute signal SMUT and the operation stop signal NLSP. To do. This allows
The audio noise reduction circuit 61 starts an audio noise reduction operation (range bit replacement or synchronization signal replacement), and the mute circuit 51g releases mute.
The audio data output from the companding decoding unit 51f is input to the DA converter 51h.

(D) Overall operation FIG. 4 is an explanatory view of the overall operation of the audio signal processing device of the present invention. When the radio wave condition is good and the CN ratio is large (condition), the number of error flags is less than or equal to the second set value TH2, and since the synchronization signal is detected, the voice noise reduction circuit 61 reduces the voice noise. No action is taken. Also, audio mute is not performed. When the radio wave condition deteriorates (the CN ratio deteriorates) and the number of error flags becomes equal to or larger than the first set value TH1 (condition), the audio noise reduction circuit 61 starts the audio noise reduction operation and performs the range bit replacement operation. . Also, if out-of-sync is detected, the sync signal bits are replaced. In this state, audio mute is not performed and noise-reduced audio is output.

When the radio wave condition further deteriorates and the mode changes from the previous mode to a different mode, in other words, when the erroneous recognition of the mode occurs (state), the correct mode is instructed to the DA converter 51h thereafter. become unable. Therefore, the audio mute controller 71c outputs the audio mute signal SMUT (* SMUT means an inverted signal of the SMUT) for audio mute, and outputs the operation stop signal NLSP to output the audio noise reduction circuit 61. Stop the operation. In this state, the radio wave condition becomes good, and the radio wave condition is improved a predetermined number of times (= 36 times) or more, that is, 3
When the sync signal is detected (state) for 6 ms, the audio mute signal SMUT and the operation stop signal NLSP are set to the low level. As a result, the audio noise reduction circuit 61 starts the audio noise reduction operation (range bit replacement or synchronization signal replacement), and the mute circuit 51.
g releases the mute and outputs sound. Then, the radio wave condition further improves and the number of error flags becomes the second set value T.
When H2 or less (state), the audio noise reduction circuit 61
The noise reduction operation of is finished.

(E) Modified example In the above, when the radio wave condition deteriorates (the CN ratio deteriorates) and the number of error flags exceeds the first set number, the range bits are replaced, and This is a case where the sync noise is detected by detecting the loss of synchronism to reduce audio noise. That is, this is the case where the range bit and the synchronization signal bit are replaced independently. However, the radio wave condition deteriorates and the CN ratio deteriorates,
When the number of error flags exceeds the first set number, the range bit and the sync signal bit are replaced at the same time (voice noise reduction operation), and when the CN ratio is improved, the range bit and the sync signal bit are replaced at the same time. It can also be configured to stop. FIG. 5 is a configuration diagram of such a modification, and the difference from FIG. 2 is that the line for inputting the out-of-synchronization detection signal SYO to the operation control unit 61e is deleted.

In FIG. 5, the operation control section 61e indicates that the radio wave condition has deteriorated and the number of error flags is the first set value TH1.
When the above is reached, the range bit in the bit stream is controlled to be replaced at the range bit replacement timing with the range data output from the range data generating section 61g, and the synchronization signal bit in the bit stream is synchronized with the synchronization signal generating section 61f. When the synchronization signal output timing is changed so that it is changed at the synchronization signal change timing, and the radio wave condition is improved and the number of error flags becomes equal to or less than the second set value TH2, the range bit and the synchronization signal bit are changed at the same time. Control to stop. Although the present invention has been described above with reference to the embodiments, the present invention can be variously modified according to the gist of the present invention described in the claims, and the present invention does not exclude these.

[0037]

As described above, according to the present invention, an error in a predetermined bit in a control signal (control code), for example, an error in a mode bit in A mode / B mode is detected to mute the sound. The voice can be output until the state becomes bad and the A mode / B mode changes, that is, the timing for applying the voice mute can be extended as much as possible, which makes it possible to prolong the time during which the voice is output, and frequently. It is possible to prevent the situation where the audio is muted. Further, according to the present invention, the audio mute is canceled when the sync signal is detected a predetermined number of times or more, for example, the number of frames used for majority decision, so that the mode is surely released when the audio mute is released. Is specified, and it is possible to prevent generation of abnormal noise due to indeterminate mode.

[Brief description of drawings]

FIG. 1 is a configuration diagram of an audio signal processing device of a BS tuner of the present invention.

FIG. 2 is a configuration diagram of an audio noise reduction circuit.

FIG. 3 is a configuration diagram of an audio mute control unit.

FIG. 4 is an explanatory diagram of the overall operation.

FIG. 5 is a configuration diagram of an audio noise reduction circuit according to a modified example of the present invention.

FIG. 6 is a frame configuration diagram of an audio signal.

FIG. 7 is an explanatory chart of control codes.

FIG. 8 is a chart for explaining specifications of a voice encoding system.

FIG. 9 is an explanatory diagram of range bits.

FIG. 10 is an explanatory diagram of quasi-instantaneous compression.

FIG. 11 is an explanatory diagram of interleaving.

FIG. 12 is a block diagram of a transmitting side of satellite broadcasting.

FIG. 13 is a configuration diagram of a conventional satellite broadcast receiving device.

FIG. 14 is a configuration diagram of a conventional audio signal processing unit.

[Explanation of symbols]

 51..Sound signal processing unit 61..Sound noise reduction circuit 71..Sound mute control unit

Claims (5)

[Claims] 1. A demodulation of a voice signal including a sync signal, a control code, a range data, and a data-compressed voice data for M channels and N samplings for each frame, and error detection and correction processing is performed on the demodulated bit stream. In the audio signal processing device of the BS tuner, which outputs the audio data in the error-corrected bit stream and decompresses the audio data in the bit stream, and inputs the audio data to the DA converter, the frame synchronization signal is detected from the bit stream. A synchronization detection unit, a radio wave state monitoring unit that determines that the radio wave state has deteriorated based on the number of error detections detected by the error detection and correction process, a frame synchronization signal generation unit that internally generates a frame synchronization signal, A range data generator that internally generates predetermined range data; A replacement timing generation unit that generates a synchronization signal bit replacement timing and a range bit replacement timing, and a range bit in the bit stream when the radio wave condition deteriorates. A range bit replacement unit that replaces at the replacement timing, and a synchronization signal bit replacement unit that replaces the synchronization signal bit in the bitstream with the synchronization signal output from the synchronization signal generation unit at the synchronization signal replacement timing when out of frame synchronization is detected. , A control code monitoring section that monitors a predetermined bit in the control code and detects an error when the radio wave condition deteriorates, and outputs an audio mute signal when the predetermined code error is detected by the control code monitoring section Audio mute And control unit, the audio signal processing device BS tuner equipped with audio mute circuit to mute the sound based on the audio mute signal. 2. The audio signal processing device for a BS tuner according to claim 1, wherein the audio mute controller outputs an audio mute signal when detecting an error in the mode bit of the A mode / B mode in the control code. 3. The audio signal processing device for a BS tuner according to claim 1, wherein the audio mute control unit cancels the audio mute when the sync signal is continuously detected a predetermined number of times or more. 4. A voice signal including a sync signal, control code, range data, and data-compressed voice data for M channels and N samplings is demodulated for each frame, and error detection and correction processing is performed on the demodulated bit stream. In the audio signal processing device of the BS tuner, which outputs the audio data in the error-corrected bit stream and decompresses the audio data in the bit stream, and inputs the audio data to the DA converter, the frame synchronization signal is detected from the bit stream. A synchronization detection unit, a radio wave state monitoring unit that determines that the radio wave state has deteriorated based on the number of error detections detected by the error detection and correction process, a frame synchronization signal generation unit that internally generates a frame synchronization signal, A range data generator that internally generates predetermined range data; A replacement timing generation unit that generates a synchronization signal bit replacement timing and a range bit replacement timing, and a range bit in the bit stream when the radio wave condition deteriorates. A synchronization signal and range bit replacement unit that replaces the synchronization signal bit in the bitstream with the synchronization signal output from the synchronization signal generation unit at the replacement timing, and controls when the radio wave condition deteriorates. A control code monitoring unit that monitors a predetermined bit in the code and detects an error, a voice mute control unit that outputs a voice mute signal when the control code monitoring unit detects an error in the predetermined bit, and a voice mute signal Based on voice Audio signal processing apparatus BS tuner equipped with audio mute circuit to mute. 5. The audio signal processing device for a BS tuner according to claim 1, wherein the audio mute control unit cancels the audio mute when the sync signal is continuously detected a predetermined number of times or more.