JPH09307516A - Digital audio broadcast reception method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To avoid a sense of incongruity in terms of a listening sense by outputting an audio signal without noise even when the reception state is deteriorated. SOLUTION: A scale factor detection/storage section 65 detects and stores a scale factor included in an MPEG audio bit stream STM when the reception state is excellent. A changeover circuit 68 replaces a scale factor in the MPEG audio bit stream ATM with the stored scale factor when the reception state is deteriorated. An MPEG audio decoding section 69 uses the replaced scale factor to decode an audio signal from the bit stream STM.

Description

Detailed Description of the Invention

[0001]

[0001] 1. Field of the Invention [0002] The present invention relates to a digital audio broadcast receiving method, and more particularly, to a digital audio broadcast signal receiving method, demodulating an MPEG audio bit stream from the received signal, and allocating an allocation included in the MPEG audio bit stream. The present invention relates to a digital audio broadcast receiving method for decoding audio sampling data into an audio signal using data and a scale factor.

[0002]

2. Description of the Related Art MPEG audio is an ISO / IEC standard method for high-quality, high-efficiency stereo audio coding, which uses 32-bit sub-band coding (band division coding) for compression and is psychoacoustic. It realizes highly efficient compression by utilizing various characteristics. M in Europe
Digital audio broadcasting using PEG audio,
Studies are underway toward the realization of digital video broadcasting.

The human ear cannot hear sounds below a certain level, and a characteristic curve obtained by plotting this level for each band is called a minimum masking threshold curve (minimum audible limit curve) MTC. (See Figure 5). The masking effect changes depending on the situation of surrounding sounds, and even a sound having a level equal to or higher than the minimum masking threshold curve MTC cannot hear a small sound due to a loud sound.
This is because the masking threshold curve is shown in FIG.
This is because the sound component below the curve is masked and cannot be heard by the human ear. In consideration of the above, sounds below the masking threshold level MTC 'are not quantized, but sounds above the masking threshold level are quantized. In the case of quantization, the quantization bit number is assigned and quantized according to the size of the difference between the audio level and the masking threshold level in each subband, and the quantization data (sampling data) and the assigned bit number are set. Is output.

Specifically, the audio signal is subdivided into 32 bands and subband coding of 32 bands is performed. That is, the entire band is divided into 32 equally spaced frequency widths, and each signal is sampled at 1/32 of the original sampling frequency and encoded. In MPEG audio, 384 (= 12 × 32) samples are set as one frame, and each subband is converted into 12 samples and 32 bands for encoding. The data of 12 samples in one band is
L channel (0 channel) waveform, R channel (1
Channel) waveform and the scale factor is determined for each. That is, normalization is performed so that the maximum value of each of the twelve waveforms becomes 1.0, and the normalized magnification is encoded as a scale factor. By this encoding, it is possible to limit the generation of quantization noise only to a region where both the frequency and time are close to a large signal component,
The effect of the psychology of hearing works, and it becomes difficult to detect noise.

Further, the number of quantization bits of each subband is determined and used as allocation data. The masking effect can be used most effectively by specifying the quantization precision (quantization bit number) up to the masking level, which takes the critical bandwidth into consideration. As a result of the masking, a band containing only a signal of a level not recognized by the auditory system can completely lose information. In such a case, no bit is assigned as sample data (no sampling data). Each channel (0
Allocation data indicates the number of quantized bits of sample data of channel, channel 1. When the number of quantized bits = 0, no sampling data exists.

FIG. 6 is a diagram for explaining the structure of one frame of an MPEG audio bit stream.
o Access Unit). AAU is one by one
It is the minimum unit that can be decoded into one audio signal, and always contains a fixed number of samples = 384 samples of data.
The AAU includes a 32-bit header section 1, an error check code (optional) 2, and an audio data section 3, and the audio data section 3 includes allocation data 3
a, a scale factor 3b, and sample data 3c. The header section 1 includes a 12-bit all "1" synchronization word 1a, an ID "1b" that is always "1", other layers 1c, a bit rate index, a sampling frequency, mode, and other information. The audio data section 3 has a structure as shown in FIG. That is,
The allocation data 3a includes subbands sb (0 to 3).
1) indicates the number of quantized bits of 12 pieces of sampling data in each channel ch (0 channel, 1 channel), and the scale factor 3b indicates the respective normalization magnifications when the number of allocation bits is other than 0. Subband Sb with non-zero allocation bits, channel ch
Each sampling data of the corresponding scale factor S
ij is multiplied and quantized by the number of allocation bits to form sample data 3c.

FIG. 8 is a block diagram of a digital satellite broadcast transmitter. The MPEG video coding circuit 11 and the audio coding circuit 12 respectively compress the audio signal and the video signal of a predetermined program by MPEG2, and the multiplexing circuit 13 multiplexes these compressed data and additional data such as program contents. The multiplexing circuit 14 further multiplexes coded data obtained by multiplexing each program into multiple channels,
Input to the satellite channel adapter 15. The satellite channel adapter 15 scrambles the multi-channel data, performs Reed-Solomon coding, interleaves to cope with burst errors, performs convolutional coding, and finally QPSK-modulates and inputs to the transmission unit 16. To do. The transmission unit 16 up-converts to a predetermined transmission frequency and transmits from the antenna.

FIG. 9 is a block diagram of a digital satellite broadcast receiver. The QPSK demodulation unit 21 performs QPSK demodulation processing on a reception signal of a predetermined channel input through a reception circuit (not shown) to demodulate a baseband bit stream, and a clock reproduction unit 22 reproduces a bit clock (internal clock). To do. The deinterleave / descramble circuit 23 executes an operation (deinterleave, descrambling process, etc.) opposite to the transmission, and inputs the processing result to the video / audio separation circuit 24. The video / audio separation circuit 24 is an MPEG
It is separated into an audio bit stream and an MPEG video bit stream and input to an MPEG video decoding circuit 25 and an MPEG audio decoding circuit 26, respectively.
The MPEG video decoding circuit 25 reproduces a video signal from the MPEG video bitstream and inputs it to the video equipment, and the MPEG audio decoding circuit 26 decodes the audio signal from the MPEG audio bitstream and inputs it to the audio equipment. That is, the MPEG audio decoding circuit 26 decodes into PCM audio data using the allocation bit number, scale factor, and sample data, DA-converts the PCM audio data, and outputs it.

[0009]

If the CN ratio (Carrier to Noise ratio), which is the ratio of the carrier signal to the noise signal contained therein, deteriorates, that is, if the reception condition deteriorates, it becomes impossible to detect the synchronization word or the allocation bit. Incorrect detection of numbers and scale factors. If the sync word cannot be detected, the audio signal cannot be correctly decoded, and noise occurs. Further, when a scale factor detection error occurs, a small sound is decoded into a loud sound to generate noise, and when a detection error of the allocation bit number occurs, quantization noise occurs. In view of the above, an object of the present invention is to provide a digital audio broadcast receiving method capable of outputting a noise-free audio even when the reception state is poor and eliminating discomfort in hearing.

[0010]

According to the present invention, there is provided a means for storing a scale factor included in an MPEG audio bitstream when a reception state is good, and a storage means for storing the scale factor when the reception state is bad. Means for decoding the MPEG audio bitstream into an audio signal using a certain scale factor.

[0011]

BEST MODE FOR CARRYING OUT THE INVENTION

(A) Overall Configuration FIG. 1 is a configuration diagram of a digital satellite broadcast receiver of the present invention. Reference numeral 51 is a satellite broadcasting antenna, 52 is a QPSK demodulation unit, which applies QPSK demodulation processing to a satellite broadcasting reception signal of a predetermined channel output from a reception unit (not shown) and outputs a baseband bit stream.
Is a clock reproducing unit having a PLL configuration, which reproduces a bit clock (internal clock) BCL included in the bit stream and outputs a PLL unlock signal ULK. Reference numeral 55 is a deinterleave / descramble circuit, which performs an operation (deinterleave, descramble processing, etc.) opposite to that of the transmitter on the bit stream output from the QPSK demodulation circuit, and outputs it. MPEG audio bitstream and MPEG
A video / audio separation circuit for separating into a video bit stream, 57 is an MPEG video decoding circuit for reproducing a video signal from the MPEG video bit stream and inputting it to a video device, and 58 is for decoding an audio signal from the MPEG audio bit stream. It is an MPEG audio decoding circuit that is input to an audio device.

(B) Clock Recovery Unit In the clock recovery unit 54, 54a is a phase comparison circuit,
Reference numeral 54b is a voltage V that oscillates at a frequency proportional to the control voltage output from the phase comparison circuit and outputs a bit clock BCL.
CXO (voltage controlled oscillator), 54c is a frequency dividing circuit for dividing the oscillation signal (bit clock signal) input from the VCXO and inputting it to the phase comparison circuit 54a, and 54d is QPSK
A sampling and holding circuit that samples and holds the detection output (beat stream) from the demodulation circuit 53,
Reference numeral 54e is a comparator that compares the control voltage output from the phase comparison circuit with a set voltage (reference value) and outputs a PLL unlock signal ULK when the control voltage exceeds the set voltage.

The phase comparison circuit 54a divides the frequency signal oscillated and output from the VCXO 54b, and QPSK.
A phase comparison is performed with a signal obtained by sampling / holding the detection output, and a control voltage corresponding to the phase difference is generated. The control voltage is fed back to the VCXO, and the VCXO controls the oscillation frequency by the control voltage so that the phase difference becomes zero.
Thereafter, this feedback control operation is repeated, and finally, the VCXO 54 reproduces and outputs the bit clock BCL having the same oscillation frequency as the internal clock included in the bit stream. When the reception state deteriorates and the CN ratio decreases, the control voltage value increases and exceeds the reference value, and the phase difference cannot be controlled (PLL unlocked state). The comparator 54e compares the control voltage value with the reference value, and when the control voltage value> the reference value, the PLL
The unlock signal ULK is output.

There is a relationship as shown in FIG. 2 between the CN ratio and the number of generated PLL unlock signals. When the CN ratio decreases, the number of times the PLL unlock signal is generated increases, and when the CN ratio increases, the PLL unlock signal occurs. The number of signal generations is reduced. Therefore, it is possible to determine the deterioration of the radio wave condition by counting the number of times the PLL unlock signal ULK is generated per arbitrary time.

(C) MPEG Audio Decoding Circuit (c-1) Configuration FIG. 3 is a configuration diagram of the MPEG audio decoding circuit 58, and 61 is an MPEG audio bit stream ST.
A sync detection unit for detecting a sync word from M, and 62 is an in-frame bit position output unit having a built-in counter. The counter has the same capacity as the number of AAU bits in one frame, is set to a predetermined value (for example, 0) by detecting a synchronization word, counts a bit clock BCL, and monitors a bit position in a frame based on the count value. ,Output. 63 is a header (sync word / CD
Header detection / storage unit for detecting and storing
Is an allocation detecting / storing unit that detects and stores allocation data, 65 is a scale factor detecting / storing unit that detects and stores a scale factor, 66 is a 12-bit all "1" sync word and 1-bit "1" It is a sync word / ID generator that generates an ID of ".

67 is a control unit having a microcomputer configuration, 68 is a switching circuit, and 69 is an MPEG audio decoding unit. Control unit 6
Reference numeral 7 determines the quality of the reception state based on the presence or absence of the PLL unlock signal ULK, and when it detects that the reception state has deteriorated, it controls the switching circuit 68 to control the synchronization word / ID, the header, the allocation data, and the scale factor. Control replacement. That is, if the reception state is good, the control unit 67 determines that the MPEG audio bitstream ST
The switching circuit 68 is instructed to select M. However, when the reception state deteriorates, the control unit 67 determines each unit 63 based on the bit position output from the intra-frame bit position output unit 62.
The sync word / ID, the header, the allocation data, and the scale factor replacement timing signals stored in the memory cells 66 to 66 are sequentially input to the switching circuit 68.

The switching circuit 68 inputs the bit stream STM as it is to the MPEG audio decoding unit 69 when the reception state is good and the selection of the bit stream STM is instructed. However, when the reception state is deteriorated and the replacement timing signal of each data is input from the control unit 67, the latest corresponding data that is detected and stored when the reception state is good is selected and output. That is, when the synchronization word / ID replacement timing signal is input from the control unit 67, the switching circuit 68 uses the synchronization word / ID output from the synchronization word / ID generation unit 66 to change the synchronization word / ID in the bitstream. Replace.
Further, the switching circuit 68 receives the header replacement timing signal from the control unit 67 and then receives the header detection / storage unit 6
Header stored in 3 (excluding sync word / ID)
When the header in the bitstream is replaced with, and the allocation replacement timing signal is input from the control unit 67, the allocation data stored in the allocation detection / storage unit 64 is used to replace the allocation data in the bitstream, and the scale factor replacement timing. When the signal is input from the control unit 67, the scale factor of the bitstream is replaced with the scale factor stored in the scale factor detection / storage unit 65. The sample data in the bit stream STM is input to the MPEG audio decoding unit 69 as it is.

The MPEG audio decoding unit 69 establishes synchronization by detecting a synchronization word, and thereafter extracts allocation data (number of allocation bits), scale factor, sample data, etc. from the beat stream,
PCM audio data is restored from these and the PCM
The audio data is DA converted and output.

(C-2) Operation When the reception condition is good, the MPEG audio decoding unit 69 correctly detects the synchronization word / ID, the header and the allocation data from the MPEG audio bit stream STM output from the video / audio separation circuit 56. , Scale factor, sample data can be detected and
The CM audio data is decoded, and the PCM audio data is DA converted and output. However, if the reception condition deteriorates, the sync word / ID, header, allocation data, and scale factor cannot be detected correctly, and large noise is generated.

Therefore, when the reception condition is good, the detection / storage units 63 to 65 detect and store these data.
Then, when the reception condition deteriorates, the bit stream ST
The synchronization word / ID, header, allocation data, and scale factor in M are
The ID, the header, the allocation data, and the scale factor that are detected and stored when the reception is good are replaced. When the reception condition deteriorates, the MPEG audio decoding unit 69 detects the replaced synchronization word / ID, allocation data (number of allocation bits), scale factor, and sample data from the beat stream, and uses them to generate PCM audio data. It decodes, converts the PCM audio data into an analog audio signal, and outputs the analog audio signal. By doing so, even if the reception condition deteriorates, the MPEG audio decoding unit can surely detect the synchronization word / ID, and immediately before the reception condition is good as allocation data (number of allocation bits) and scale factor. The value can be used for voice output and noise can be reduced.

(D) Modification In the above, the scale factor when the reception state is good is stored, and when the reception state is deteriorated, the scale factor in the bit stream is replaced with the latest scale factor stored. However, the configuration may be as follows. That is, the coefficient is set in advance so that the value decreases as the CN ratio decreases (the degree of deterioration of the reception state increases), and when the reception state deteriorates, the stored scale factor or The detected scale factor is multiplied by a coefficient according to the CN ratio, the obtained value is used as a scale factor, and the scale factor in the bitstream is replaced with the scale factor.

FIG. 4A is a block diagram of the scale factor detection / storage unit 65 in such a case, where 65a is a scale factor detection unit, 65b is a storage unit which stores the latest detected scale factor, and 65c is a CN. Coefficient / CN ratio correspondence table for storing correspondence between ratio and coefficient value, 65d
Is a CN ratio determination unit that determines and outputs the CN ratio based on the number of times the PLL unlock signal ULK is generated per set time, 6
Reference numeral 5e is a calculation unit that multiplies the latest scale factor detected by a coefficient corresponding to the CN ratio and outputs the result. CN ratio and P
Since the number of generated LL unlock signals has the relationship shown in FIG. 2, the CN ratio determination unit 65d can detect the CN ratio by monitoring the number of generated PLL unlock signals. By doing so, the scale factor noise can be reduced and the voice can be output when the CN ratio is relatively good, and the scale factor noise can be surely reduced and the voice can be output even when the CN ratio is bad. .

Incidentally, in the coefficient / CN ratio correspondence table 65c, as shown in FIG. 4B, the coefficients of the respective sub-bands are individually stored in correspondence with the CN ratio. Noise can be more effectively reduced by setting the coefficient of the sub-band of 3 to 5 KHz or more, which is easily noticed by human ears, among the 32 sub-bands to be smaller than the others. In the above, both the allocation data and the scale factor are replaced, but only the scale factor may be used. As described above, the present invention has been described with reference to the embodiments. However, the present invention can be variously modified in accordance with the gist of the present invention described in the claims, and the present invention does not exclude these.

[0024]

As described above, according to the present invention, the scale factor included in the MPEG audio bitstream when the reception state is good is stored, and when the reception state is bad, the stored scale factor is stored. Using MPE
Since the audio signal is decoded from the G audio bit stream, even if the reception condition becomes poor, noise-free audio output can be performed, and discomfort in hearing can be eliminated.

[Brief description of drawings]

FIG. 1 is a configuration diagram of a digital satellite broadcast receiver of the present invention.

FIG. 2 is an explanatory diagram of a relationship between a PLL unlock signal generation number and a CN ratio.

FIG. 3 is a configuration diagram of an MPEG audio decoding circuit of the present invention.

FIG. 4 is a configuration diagram of a scale factor detection / storage unit.

FIG. 5 is a masking threshold characteristic diagram.

FIG. 6 is a structural explanatory view of AAU.

FIG. 7 is a configuration diagram of an audio data unit in AAU.

FIG. 8 is a schematic configuration diagram of a digital satellite broadcast transmitter.

FIG. 9 is a block diagram of a conventional digital satellite broadcast receiver.

[Explanation of symbols]

 58 ... MPEG audio decoding circuit 61 ... Sync detection unit 62 ... In-frame bit position output unit 63 ... Header detection / storage unit 64 ... Allocation detection / storage unit 65 ... Scale factor detection / storage unit 66 ... Synchronous word / ID generation unit 67 ... Control unit 68 ... Switching circuit 69 ... MPEG audio encoding unit ULK ... PLL unlock signal

Claims (1)

[Claims] 1. A digital audio broadcast signal is received,
In a digital audio broadcasting receiving method of demodulating an MPEG audio bit stream from a received signal and decoding audio sampling data into an audio signal using a scale factor or the like included in the MPEG audio bit stream, when the receiving state is good Of the MPEG audio bitstream is stored, and when the reception condition becomes poor, the MPEG audio bitstream is decoded into an audio signal using the stored scale factor. Digital audio broadcasting reception method.