KR940000449B1 - Pcm sound signal multiplexing and out-put system - Google Patents

Abstract

The system comprises sample and holders (2a,2b), A/D converters (3a,3b), coded interleave scrambles (4a,4b), 1st multiplexer (6), a parallel-serial converter (7), an interpolation control code generating circuit (11), 2nd multiplexer (9), an audio signal separation section (20), a data interpolator (29), a control code separator (21), a sample and holder (23), a data decoder (24), a serial- parallel converter (25), a field judging section (26), deinterleave descramble error correction sections (28a,28b), a data correction section (29), and D/A converters (30a,30b).

Description

PCM voice signal multiplexing and output system

1 is a block diagram of a system of the present invention.

* Explanation of symbols for main parts of the drawings

1a, 1b, 8, 22, 31a, 31b: Low pass filter 2a, 2b, 23: Sample and holder part

3a, 3b: A / D converter 4a, 4b: coded interleaved scrambler

5a, 5b: time base compression unit 6,9: multiplexing unit,

7: Parallel-Serial Converter 10: Switch

11: interpolation control code generator 20: voice data separation unit

21: control code separation unit 24: data decryption unit

25 series-parallel conversion unit 26 field determination unit

27a, 27b: time base extension

28a, 28b: deinterleaved descramble error correction unit

29: data correction unit 30a, 30b: D / A conversion unit

The present invention relates to a PCM audio signal multiplexing and output system for time division multiplexing an audio signal to a video signal and separating a digital audio signal multiplexed into the video signal.

Existing digital audio signal multiplexing system encodes, interleaves, and scrambles digitalized R, L two-channel audio signals through A / D converter at the same time and compresses them in time-base compression, then multiplexes them in the vertical blanking period (vertical blanking time) Record on the recording medium.

Normally, a TV screen is made with 30 frame scans per second, and there are 525 scan lines per frame scan making one screen.

Here, each frame scan is composed of two field scans, and the first field scan (262.5 scan lines) serving as the first field scan and the second field scan (262.5 scan lines) serving as the second field scan each have an image display period ( That is, the first and second vertical blanking periods other than the image data storage period) exist, and audio signal data of the digitalized RL (or LR) channels are recorded in these first and second vertical blanking periods, respectively.

In the case of reproducing the audio signal multiplexed with the video signal on the recording medium, the digital data of the audio signal is read from the recording medium to correct the time base extension, deinterleave, descramble and error, and then through the D / A converter. The signal is being output.

At this time, the error correction of the voice data is corrected according to the error degree through the correction method such as the average value interpolation method, pre-wound method, muting method, etc. In the conventional voice signal multiplexing system based on the content of the data before and after the interpolation Because of this, it is not possible to cope with the large-scale outbreak error defined by the interleaved method, and since the audio data of the time-division multiplexed R, L channel is recorded at the adjacent position in time, correction is performed by mutual data in the case of relatively large outbreak error. There is a problem that cannot be done.

SUMMARY OF THE INVENTION The present invention has been made in view of the above-mentioned problems, and an object of the present invention is to provide a PCM audio signal multiplexing and output system that enables the expansion of error correction capability due to a large unexpected error.

To this end, in the present invention, an error in digital audio data is recorded in time-division multiplexing of R and L channel digital audio signals in the first and second vertical blanking periods between the first and second field scanning periods in which video signals are recorded. The interpolation control code for correction is inserted and multiplexed. During reproduction, when an uncorrectable error occurs in digital audio data of one channel, data correction is performed based on data of another channel based on the interpolation control code. It features.

Hereinafter, the present invention will be described with reference to the accompanying drawings.

FIG. 1 is a circuit block diagram of a system of the present invention, in which an audio signal multiplexing portion for digitalizing and multiplexing an analog audio signal of an L and R channel and a digital audio signal multiplexed by the audio signal multiplexing portion are output. It is divided into audio signal output part.

The voice signal multiplexing portion includes: low pass filter units 1a and 1b for removing unwanted signals included in voice signals of L and R channels, and analog voice signals through the low pass filter units 1a and 1b. Sample and holder parts 2a and 2b for sampling the A / D converters 3a and 3b for converting the analog audio signal sampled and output from the sample and holder parts 2a and 2b into digital signals. And compressing the audio signal output data of the encoded interleaved scramblers 4a and 4b for encoding the voice digital signal output from the A / D converters 3a and 3b, and the encoded interleaved scramblers 4a and 4b. The digital audio data of the L channel compressed by the time base compression units 5a and 5b and the time axis compression unit 5a are subjected to the first vertical blanking time (the vertical blanking time of the first field video signal) of the video signal. And digital of the R channel compressed by the time base compression unit 5b. The sex data is outputted in parallel with the first multiplexer 6 and the first multiplexer 6 for multiplexing alternately with the second vertical blanking time of the video signal (the vertical blanking time of the second field video signal). Parallel-to-serial converting unit 7 for converting the signals in series and transmitting the multiplexed L and R channel audio data of the parallel-serial converting unit 7 through low pass filter 8 to a video signal. A second multiplexer 9 multiplexed to the first and second vertical blanking portions of the two-field video signal, and the correlation part of the digital audio data between the L and R channels (strictly correlating the L and R channel audio data). The interpolation control code is generated with 1-bit identification code information (flag: flag is "1" if the correlation is correlated, flag "0" if the correlation is not correlated), and the second frame is generated for each voice data frame. It is provided to the multiplexer 9, and whether or not such 1-bit erasure code information (flag) is generated It consists of the interpolation control code generating circuit 11 having a switch 10 for setting.

On the other hand, the audio signal output portion is audio digital data and identification code information of L and R channels multiplexed from the output of the second multiplexer 9 to the first and second vertical blanking portions of the first and second fields of the video signal, respectively. Voice data separation unit 20 for separating a flag and control code separation unit 21 for separating identification code information among voice digital data and identification code information output from the voice data separation unit 20 separately. And a low pass filter 22 and a sample and holder unit 23 for filtering and sampling a voice digital data signal output from the control code separating unit 21, and sampling of the sample and holder unit 23. A data decoding unit 24 for decoding voice data from the output, a serial-parallel conversion unit 25 for converting and outputting the decoded voice digital data serial output of the data decoding unit 24 in parallel, and the serial-parallel conversion Voice digital de of part (25) The field judgment unit 26 determines the field from the data parallel output and transmits the fields to the respective time base extension units 27a and 27b, and deinterleaves and descrambles the outputs of the respective time axis extension units 27a and 27b. And the L and R channels output from the deinterleaved descramble error correction units 28a and 28b for performing error correction and generating a flag for whether or not error correction is possible. If an error correction impossible flag is present in the voice digital data, data correction is performed for each channel according to the average value interpolation method, the electronic holder method, or the muting method according to the 1-bit identification code information of the control code separation unit 21. If there is no error correction impossible flag in the channel audio digital data, the data correction unit 29 transmits the audio digital data to the D / A converters 30a and 30b of the next stage, and the D And low pass filters 31a and 31b for filtering voice signals converted into analog signals by the / A converters 30a and 30b.

Referring to the operation and effects of the present invention configured as described above are as follows.

First, when the audio signals of the L channel and the R channel are input to each of the low pass filters 1a and 1b, the unnecessary signal is removed therefrom. Subsequently, the L and R channel audio signals sampled by the respective sample and holder sections 2a and 2b are converted into digital signals by the A / D converters 3a and 3b to the encoded interleaved scramble sections 4a and 4b. Is provided.

The voice digital signal input to the encoding interleave scrambler 4a, 4b is coded and encrypted here and then transmitted to the first multiplexer 6 via the respective time base compression units 5a, 5b.

The L channel audio digital data among the audio digital data input to the first multiplexer 6 is time-division multiplexed at a first vertical blanking time, which is a vertical blanking time of the first field video signal, and the R channel audio digital data is second-segmented. Time division multiplexing is performed on the second vertical blanking time, which is the vertical blanking time of the field video signal. The L / R channel audio digital data multiplexed by the PCM in the first multiplexer 6 is transferred to the second multiplexer 9 through which the video signal is input through the parallel-serial converter 7 and the low pass filter 8. Is sent.

Accordingly, the second multiplexer 9 alternately multiplexes the voice digital data of the L channel and the R channel in the first and second vertical blanking periods existing in each of the first and second field scanning periods of the video signal.

At this time, when the switch 10 of the interpolation control code generation circuit unit 11 is turned on, the interpolation control code generation circuit unit 11 compares the correlation between the L channel and the R channel voice digital data, and if the correlation between the two channels is high, it is 1 bit. If the flag of identification code information is "1" and the correlation of both channels is low, the flag is generated "0" and multiplexed to the video signal with audio digital data for each field frame.

When the voice digital data multiplexed in the first and second vertical blanking periods of the first and second field scanning periods of the video signal are transmitted to the voice data separation unit 20 of the audio signal output unit, the audio data separation unit 20 The video signal and the audio signal are separated to transmit only the audio digital data and the interpolation control code multiplexed thereto to the control code separation unit 21. Accordingly, the control code separation unit 21 is separated from the voice digital data into an interpolation control code, that is, a 1-bit identification code (flag), and transmitted to the data correction unit 29, and the remaining pure voice digital data is passed through the low pass filter 22. And the sample and holder section 23 are transmitted to the data decoding section 24. The voice digital data input to the data decoder 24 is decoded and then transmitted to the field determiner 26 via the serial-to-parallel converter 25.

The digital data separated by the field determining unit 26 into the L channel and R channel voice signals are transmitted to the deinterleaved descramble error correcting units 28a and 28b through the time base extension units 27a and 27b, respectively. The deinterleaved descramble error correction portions 28a and 28b are restored to the original voice digital data and error correction is performed. At this time, if there is voice data that cannot be corrected in error, the deinterleaved descramble error correctors 28a and 28b indicate that the error correction impossible flag (1 bit identification information: "1" means error correction impossible, and "0" indicates error correction possible). Is transmitted to the data correction unit 29 together with the voice digital data generated and reconstructed.

The data correction unit 29 receives voice digital data of the L channel and the R channel and the error correction impossible flag from the deinterleaved descramble error correcting units 28a and 28b. It is determined that error correction of the data is completed, and the voice digital data of each channel is transmitted to the D / A converters 30a and 30b of the subsequent stages as they are.

However, if the error correction impossibility flag output from the deinterleaved descramble error correction units 28a and 28b is "1", the data correction unit 29 determines that a burst error that cannot be error correction has occurred in the voice digital data. The interpolation control code (1 bit identification code: flag) inputted from the control code separation unit 21 is read. If the flag of the interpolation control code is "1", it is judged to be highly correlated to the voice digital data of both channels and is different. The error is corrected by substituting the audio digital data of the channel.

If the flag of the interpolation control code at this time is "0", it is determined that the correlation is low in the voice digital data of both channels, and the error is corrected through the average value interpolation method, the pre-hold method, or the muting method, which are general error correction techniques.

The voice digital data error corrected by the data correction unit 29 is converted into an analog voice signal by the D / A converters 30a and 30b and then passed through the low pass filters 31a and 31b to the voice of the L channel and the R channel. It is output as a signal.

As described above, the present invention is time-division multiplexed on a video signal by digitalizing a two-channel audio signal, and during playback, when an uncorrectable unexpected error occurs in digital audio data, correction is performed with data of a corresponding channel, so that a large unexpected error occurs. Even if is generated, a unique effect of allowing correction is obtained.

Claims (2)

Sample and holder sections 2a and 2b and A / D converters 3a and 3b for sampling and converting analog audio signals of L and R channels into digital signals, and the A / D converters 3a and 3b. Encoding interleaved scrambles 4a and 4b for coding and encrypting the voice digital signal of the first and second audio data of the encoded interleaved scramblers 4a and 4b through the time base compression units 5a and 5b. The audio digital data of the first multiplexer 6 which is time-division multiplexed according to the 2 vertical blanking times and serially transmitted through the parallel-to-serial converting unit 7 and the first multiplexer 6 is the first signal of the video signal. In the 2 vertical blanking periods, flags corresponding to the high and low L and R channel correlation degrees of the audio digital data obtained by the interpolation control code generation circuit unit 11 are alternately multiplexed for each of the first and second vertical frame units of the video signal. The second multiplexer 9 and the multiple outputs of the second multiplexer 9 Separates the audio data separation unit 20 for separating and removing the video signal, and the control code for separating the interpolation control code (flag) from the output of the audio data separation unit 20 and providing the correction code to the data correction unit 29. A section 21, a sample and holder section 23 and a data decoding section 24 for sampling and data decoding pure voice digital data output from the control code separation section 21, and a serial-to-parallel conversion section ( A field decision unit 26 for determining the first and second vertical fields according to the voice digital signal of the data decoding unit 24 which has passed through 25, and transmitting the first and second vertical fields to the respective time base extension units 27a and 27b; Deinterleaved descramble error correction unit 28a, 28b that receives a voice digital signal passing through (27a, 27b) and generates an error correction impossible flag (identification code) when a burst error occurs through deinterleaving, descramble, and mutual channel data replacement. And the deinterleave Data correction unit for receiving L, R channel voice digital signals and error correction impossible flags from the drive descramble error correction units 28a and 28b and selectively performing L and R channel voice data correction with or without the error correction impossible flag ( 29) and a D / A converter (30a, 30b) for converting the voice digital signal passed through the data corrector (29) into an analog signal. The data correction unit 29 outputs the input L and R channel voice digital signals as they are, if there is no error correction flag at the output of the deinterleaved descramble error correction units 28a and 28b, and deinterlace descramble error correction unit. If an error correction flag is present in the output of (28a, 28b), the interpolation control code flag input from the control code separation unit 21 is judged. When the data of channel is replaced by frame unit and the flag "0" having low correlation between both channels is used, error data of the L and R channels is replaced by one of the normal pre-holding method, average interpolation method, and muting method. PCM audio signal multiplexing and output system, characterized in that for correcting.