JP2000347697A - Voice record regenerating device and record medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To easily detect stored voice level even if the voice data of a sub-band is not decoded by detecting a voice signal or the voice level of voice data, and adding data showing detected voice level to an assistant data part of a frame format. SOLUTION: An encode/decode part 110 performs compression process of input voice data, and converts into the signal of specified frame format. A system microcomputer 103 controls this operation, and controls voice data to be compression/expansion processed to be temporarily stored and read out. When the encode/decode part 110 compresses and converts a voice signal, the system microcomputer 103 adds the data of the voice level of the voice signal to the assistant data part of the specified frame format, and further at reading out the signal, the data of the voice level added to the assistant data part of the specified frame format is detected to change the gain of electronic volume 108 according to the height.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

The present invention relates to an audio recording / reproducing apparatus for recording / reproducing audio compressed data and a recording medium.

[0002]

2. Description of the Related Art MPEG (Moving Picture Exclusive) is a typical audio compression method for reducing the amount of information by using the psychoacoustic characteristics and omitting the amount of information with low sensitivity to hearing.
pertsGroup) There is audio. Three types of audio compression processing methods, MPEG1, MPEG2 and MPEG4, have been standardized for MPEG audio.
There are three types of audio compression modes: yer1, Layer2, and Layer3.

The MPEG1 audio layer 1 audio compression processing method uses 384 samples of audio data as one processing unit to convert an input audio signal into three different frequency bands.
2 sub-bands (hereinafter referred to as SBs), quantized, and normalized by a factor such that the maximum amplitude of the sample data waveform of each sub-band is approximately 1.0. (hereinafter, referred to as SF) and bit allocation that appropriately allocates bits to each SB.

[0004] Layer 2 of MPEG1 audio is layer 1
In addition to the audio compression processing described above, audio data of 384 × 3 samples are used as one processing unit, and bit allocation is performed using a compression processing table prepared for each data transfer rate for a plurality of data transfer rates. This is a voice compression method that performs voice coding with high quality and high efficiency.

[0005] Layer 2 of MPEG2 audio is basically an extension of the stereo of MPEG1 to multi-channels, or a structure in which the sampling frequency is reduced to 1/2 so that data can be transmitted at a low speed. The method uses the same technology as MPEG1.
Therefore, description of MPEG2 is omitted.

[0006] Layer 3 of MPEG1 audio is Layer 2
In the processing of the audio compression method, the variable discrete cosine transform, which is a frequency division method that is unlikely to cause aliasing, and the entropy that makes the code length variable, focusing on the bias in the appearance probabilities of parameters appearing in the process of compression processing (Huffman) This is a speech compression method that performs encoding with higher efficiency by using processing such as encoding. An overview of MPEG audio is available in the latest MPEG textbook (first published in August 1994,
ASCII Modified Discrete Cosine Transform is described on page 176, and entropy coding is described on page 17).

There is an audio recording / reproducing apparatus which automatically changes the output level in accordance with the audio level of recorded audio data when reproducing the above-mentioned compressed data. In this case, a means for performing processing for expanding compressed audio data and then detecting the audio level is used.

[0008]

However, it is necessary to perform a decoding process for reproducing the audio data in order to detect the audio level, and a complicated process is required until the audio level is detected. The object of the present invention is
It is an object of the present invention to provide an audio recording / reproducing apparatus and a recording medium capable of easily detecting a recorded audio level without decoding the audio data.

[0009]

According to the present invention, there is provided an audio recording / reproducing apparatus for recording / reproducing an audio signal by compressing / expanding the audio signal, wherein the audio signal is divided into sub-bands of different audio frequency bands,
Band dividing means for quantizing to audio data, masking threshold calculating means for calculating a masking threshold, level comparing means for comparing the level of the audio data with the masking threshold, and a comparison result of the level comparing means Bit allocation means for allocating bits to the sub-bands, re-quantization means for re-quantizing voice data of the number of bits allocated to the sub-bands by the bit allocation means, and detecting a voice level of a voice signal or voice data. Audio level detecting means for converting the audio level detected by the audio level detecting means into a supplementary data portion of a frame format. Audio level control unit based on data indicating the level An audio recording and reproducing apparatus comprises control means for controlling the gain.

The present invention also relates to an audio recording / reproducing apparatus for recording / reproducing an audio signal by compressing / expanding the audio signal, wherein the bit allocation means calculates a scale factor of the sample data of the sub-band, and the audio level detecting section comprises: An audio recording / reproducing apparatus for detecting a maximum value of the scale factor and adding the maximum value of the scale factor to the auxiliary data section.

The present invention also relates to an audio recording and reproducing apparatus for recording and reproducing an audio signal by compressing and expanding the audio signal, wherein the audio level detecting section detects an audio level of an audio signal or audio data input to the audio recording and reproducing apparatus. The control means is a voice recording / reproducing apparatus for adding the voice level to the auxiliary data section.

The present invention also relates to a recording medium for recording a signal in a frame format in which data of a scale factor and an audio signal or audio data obtained by compressing audio data are recorded. This is a recording medium in which a maximum value or an average value of the scale factors is arranged and recorded.

[0013]

BEST MODE FOR CARRYING OUT THE INVENTION

DESCRIPTION OF THE PREFERRED EMBODIMENTS An audio recording / reproducing apparatus according to an embodiment of the present invention will be described. FIG. 1 is a block diagram showing a configuration of an audio reproducing apparatus according to one embodiment of the present invention.

The recording medium 101 is a recording medium for recording audio data that has been subjected to audio compression in the form of a signal in the frame format of the MPEG audio standard. In this embodiment, an IC memory is used as a recording medium.

The interface unit 102 includes the recording medium 1
Then, the audio data read from “01” is transferred to the encoding / decoding unit 110, and the audio data expanded by the encoding / decoding unit 110 is transferred to the audio reproduction circuit unit 112.
Further, the digital audio data input from the outside is transferred to the encoding / decoding unit 110, and the audio is compressed.
The signal is converted into a signal of a predetermined frame format by the encoding / decoding unit 110 and recorded on the recording medium 101.

The audio reproducing circuit unit 112 includes a digital filter (D / F, Digital Filter) 106 and a digital / analog converter (D / A, Digital Analog Converter) 10.
7, an electronic volume 108 and a buffer circuit 109 are provided.

The encoding / decoding unit 110 performs compression processing of MPEG1 described later on the input audio data input from the interface unit 102, and converts it into a signal of a predetermined frame format. In addition, the compressed audio data converted into a signal of a predetermined frame format is expanded.

The system microcomputer 103 performs encoding /
In addition to controlling the operation of the decoding unit 110, it also performs control such as temporarily storing and reading audio data for compression / expansion processing in the memory 111. Also, the system microcomputer 1
03, when the encode / decode unit 110 compresses the audio signal and converts it into a signal of a predetermined frame format, the data of the audio level of the audio signal is added to the auxiliary data part of the predetermined frame format, When reading out the signal of the frame format, the data of the audio level added to the auxiliary data portion is detected, and the control of changing the gain of the electronic volume 108 based on the magnitude is performed. Further, the system microcomputer 103 includes the operation unit 10
4 to control the operation of the audio recording / reproducing apparatus, and control the operation of the entire audio recording / reproducing apparatus, such as transmitting display data to be displayed on the display unit 105.

The operation unit 104 has an instruction button for instructing start and stop of recording and reproduction of the audio recording and reproducing apparatus. The display unit 105 displays the operation state of the operation unit 104, the level of the sound being recorded and reproduced, and the like.

FIG. 2 is a diagram for explaining the configuration of the encoding / decoding section of the audio recording / reproducing apparatus according to the first embodiment of the present invention.

The band dividing unit 201 is a signal processing circuit that includes a filter bank for dividing a voice frequency band into a plurality of frequency bands using a plurality of filters, and quantizes input voice data. Input audio data consists of 32 different SBs
And quantized.

The masking threshold calculation unit 202 calculates the power level of each SB by fast Fourier transform (FFT, Fast Fourier Transform) processing, and obtains a masking threshold using psychoacoustic characteristics. Masking means, for example, that in a quiet environment, the sound of a babbling can be heard, but in a storm it may not be heard. A state in which you cannot hear. The masking threshold is a maximum audio level at which an audio signal in a certain SB is masked by adjacent audio.

The bit allocation / scale factor calculation unit 203 determines the masking threshold value output from the masking threshold value calculation unit 202 and the level of the audio data of each SB, and allocates an appropriate number of bits to each SB. After allocating an appropriate number of bits to each SB, the requantization unit 204 re-quantizes the audio data with the allocated number of bits and outputs compressed audio data. Further, the bit allocation / scale factor calculation unit 203 requantizes and compresses the waveform of the sample data in the SB to the maximum amplitude value of the sample data when the waveform is normalized so that the maximum amplitude becomes approximately 1.0. A scale factor (SF) indicating the ratio of the audio level represented by the audio data to the maximum amplitude value (magnification of the compressed data) is calculated and output.
The audio data is represented as an accurate audio signal by the requantized data and the calculated SF data.

The formatting unit 205 includes a synchronization signal,
Header information including identifiers of various modes of MPEG audio, the number of bits allocated to each SB determined by the bit allocation / scale factor calculation unit 203,
The audio data re-quantized by the SF and re-quantization unit 104 is converted into a signal of a frame format of the MPEG1 audio standard and output.

The level comparing section 206 compares the level of quantization noise generated when the audio data is requantized with the number of bits allocated to each SB with a masking threshold,
Further, the difference between the signal level in each SB and the level of the masking threshold is compared with a predetermined reference value. The comparison result is sent to the bit allocation / scale factor calculation unit 203, and the bit allocation / scale factor calculation unit 203 determines the number of bits to be allocated to each SB based on the comparison result.

The input audio data input from the interface unit 102 is input to the
Is converted into audio data for each SB divided into. At the same time, the input voice data input from the interface unit 102 is input to the
FT (Fast Fourier Transform) processing is performed, and a power level is calculated for each SB. By calculating the power level for each SB, the masking threshold calculation unit 202 outputs a masking threshold of each SB due to the masking effect.

In the bit allocation / scale factor calculation unit 203, the masking threshold is
01 is compared with the signal level of the audio data in each SB, which is the output of 01, and the number of bits to be assigned to each SB is determined so as to be within a predetermined total number of assignable bits. In the processing procedure for assigning the number of bits to each SB, first, the SB having the largest dynamic range is selected from the 32 SBs. Each SB is scanned to select the SB having the largest quantization noise to masking threshold ratio (NMR), and a portion of the number of bits is assigned to the SB. The quantization noise changes according to the number of bits allocated to the SB and N
Since the MR changes, the SB is scanned again after calculating the NMR again, and the SB having the largest NMR is selected.
The process of allocating a part of the number of bits to the SB and calculating the NMR again is repeated until the allocatable bits are all allocated. afterwards,
According to the number of bits assigned to each SB, the requantization unit 204 re-quantizes and compresses the audio data assigned to each SB into audio data.

In the formatting unit 205, MPEG1
Data such as a scale factor (SF) and audio data are arranged and output so as to be a signal of a frame format determined by an audio standard.

FIG. 4 shows an audio recording / reproducing apparatus for recording / reproducing a voice by compressing / expanding the voice, and in which the input voice data is different.
FIG. 9 is a diagram illustrating audio data divided into B and assigned bits.

The vertical axis of FIG. 4 (a) shows the level of the masking threshold (hatched portion) obtained by using the audio level of the input audio data and the psychoacoustic characteristics in each SB. The horizontal axis indicates the number of each SB, with the SB0 side having a low frequency and the SB31 side having a high frequency. FIG. 4 (a)
The dynamic range is larger as the audio level of the input audio data is larger than the masking threshold, indicating that more bits are required.

FIG. 4B is a diagram showing a state when the audio data is requantized by the bit number allocated in each SB by the bit allocation / scale factor calculation unit 103. The vertical axis represents the number of bits of audio data allocated to each SB.
Is the number of quantization bits when re-quantizing the audio data. The horizontal axis indicates the SB number, with the SB0 side being a low frequency and the SB31 side being a high frequency. As shown in FIGS. 4A and 4B, as the SB of the input audio data whose audio level is higher than the masking threshold, the number of bits is allocated more.

No bits are assigned to SB26 and SB29 because the audio level is lower than the masking threshold. Further, although the sound level of the SB 27 is higher than the masking threshold, no bit is allocated to the SB 27 because the algorithm preferentially allocates bits to SBs whose voice level is higher than the masking threshold.

Next, the frame format of the MPEG1 audio standard will be described. FIG. 5 is a schematic diagram illustrating the arrangement of various data in the MPEG1 audio frame format used in the audio recording / reproducing apparatus of the present embodiment.

A synchronization signal (sync word: 1111 1111 1111) indicating the start of a frame is arranged in the header section. Error detection information is arranged in an error check section. Audio data
In the section, 32 S
Allocation indicating the order of encoding the audio data of B
(allocation) data, the scale factor (scale of the compressed data) representing the ratio when the waveform of the sample data in the SB is normalized so that the maximum amplitude becomes approximately 1.0.
factor) and audio sample data (compressed data) obtained by dividing the audio data. The multi-channel extension (mc extension) unit converts MPEG1 to M
This is an extension part for making a PEG2 multi-channel. The ancillary data part is the part used as auxiliary data.

Next, an operation of detecting an audio level in the audio recording / reproducing apparatus according to one embodiment of the present invention will be described. FIG. 6 is a flowchart illustrating an operation of detecting an audio level in the audio recording and reproducing apparatus according to the present embodiment.

First, the frame number of the frame data to be read first is set to 1, and the first frame data of the compressed audio data is read and transferred to the memory 111 for storage. (ST601), (ST602)

Next, the data in the memory storing the maximum value of the scale factor (SF) is cleared. (ST60
3) The system microcomputer 103 calculates the scale factor (S) of each SB from the frame data transferred to the memory 111.
The data of F) is sequentially read from the SB on the lower frequency side.
When the monaural mode of the MPEG1 audio layer 1 and the sampling frequency (fs) = 32 KHZ, there are 32 SF data per frame. (ST60
4), (ST605)

In the case of MPEG1 audio, SF data is in 2 dB steps from +6 dB to -118 dB. By sequentially comparing the read SF data of each SB and updating the maximum value of the SF stored in the memory, the maximum value of the SF in the frame can be obtained.
The obtained final value is used as the maximum level data of the frame. (ST606), (ST607), (ST608)

The maximum level of SF for the first frame
Is determined, 1 is added to the frame number of the read frame, the data of the next frame is read into the memory 111, and the SF of the maximum level of that frame is read in the same procedure.
Is repeated. (ST6
09), (ST610)

As shown in the above operation procedure, the maximum level in each frame can be detected by reading only the SF data without expanding the compressed audio data.

FIG. 7 is a flowchart for explaining the operation of automatically controlling the gain of the electronic volume at the time of reproduction in the audio recording / reproducing apparatus of this embodiment.

First, an instruction for detecting the audio level of the program tune is performed using an instruction button provided on the operation unit 104.
(ST701)

The system microcomputer 103 includes an operation unit 104
(SF) frame.max.n, which is data of the maximum value of SF of each frame, is searched in order from the first frame in the program music according to the instruction of the instruction button. Here, n is 1 to N
, And N is the number of frames of one music program.
(ST702)

Average value (SF) prog of maximum value data (SF) frame.max.n of SF of each frame in the program music
Calculate ram.av. (SF) program.av. Is obtained by the following equation. (SF) program.av. = Σ (SF) frame.max.n / N,
(N = 1 to N), where N is the number of frames of one program (ST703)

The average value (SF) progra obtained in ST703
m.av. is recorded in the recording information section for managing the program music in the recording medium. (ST704)

When the reproduction start button provided on the operation unit 104 is operated, the system microcomputer 103 reads the average value recorded in the recording information section for managing the program music in the recording medium, and based on the value, the electronic volume 108 Is controlled to automatically change the gain. (ST705), (S
T706)

After the gain of the electronic volume 108 is changed, the audio data is demodulated, input to the audio circuit unit 112, and output as an analog signal. (ST707)

FIG. 8 is a flow chart for explaining the operation of controlling the gain of the electronic volume when the data of the auxiliary data part of the frame format signal is not used in another audio recording / reproducing apparatus of this embodiment.

First, an instruction for detecting the audio level of the program music is made by using an instruction button provided on the operation unit 104.
(ST801)

The data (SF) i of the SF given to the 32 SBs for each frame is read and read (S
F) Find an average value (SF) frame.av.n of i (i = 0 to 31). (SF) frame.av. Is represented by the following equation. (SF) frame.av.n = Σ (SF) i / 32, (i = 0 to
31, n = n) The obtained value of (SF) frame.av.n is stored in the internal memory as the average audio level of each frame. (ST802)

Using the average audio level (SF) frame.av.n (n = 1 to N) of each frame obtained in ST802, the average value (SF) program.a over the entire program music piece
Calculate v. (SF) program.av. Is obtained by the following equation. (SF) program.av. = Σ (SF) frame.av.n / N,
(N = 1 to N), where N is the number of frames of one tune program (ST803)

The average value data (S
F) Program.av. Is recorded in the recording information section for managing the program music in the recording medium. (ST804)

When the reproduction start button provided on the operation unit 104 is operated, the system microcomputer 103 causes the average value data (SF) program.a recorded in the recording information section in the recording medium.
v. and read the electronic volume 108 based on the value.
Is controlled to automatically change the gain. (ST870
5), (ST706)

After the gain of the electronic volume 108 is changed, the audio data is demodulated, input to the audio circuit unit 112, and output as an analog signal. (ST807)

Information on the average audio level of each program music or information on the deviation of the average audio level of each program music based on the audio level of the first program music is recorded on a recording medium as recording information for managing the program music. Thereby, the audio level of the program music can be known without performing the operation of decoding the compressed audio data and checking the audio level at the time of reproduction. When the reproduction of the program music is instructed, by reading the data of the average audio level, which is the recording information of each program music, the gain of the electronic volume 308 is adjusted so that the reproduction can be performed continuously at the same level in the sense of hearing. become.

Next, an audio recording / reproducing apparatus according to a second embodiment of the present invention will be described. FIG. 3 is a diagram for explaining the configuration of the encode / decode section of the audio recording / reproducing apparatus according to the second embodiment of the present invention.

The configuration of the audio recording / reproducing apparatus shown in FIG. 3 is different from the configuration of the audio recording / reproducing apparatus shown in FIG.
a, and the other configuration is the same. The audio level detection unit 207a detects the audio level of the input audio data before the audio compression processing.

The data of the detected audio level is sent to the formatting unit 205 and added to the auxiliary data unit of a predetermined frame format.
Will be recorded. In the present embodiment, the audio level data added to the auxiliary data part of the predetermined frame format does not directly represent the audio level of the same frame, and the data of the same audio level is arranged over a plurality of frames. There is also.

At the time of reproducing the audio data, it is not possible to change the gain of the electronic volume based on the data of the average audio level of the program music obtained by using the data added to the auxiliary data section. This is the same as the first embodiment of the present invention.

The above-described audio recording / reproducing apparatus uses an IC memory as a recording medium.
D (Compact Disc), MD (Mini Disc), DVD (Digital
An optical disk, a magneto-optical disk or a phase change optical disk using a Versatile Disc (PD) or a Phase Change Disc (PD) may be used, or a magnetic hard disk may be used.

[0061]

According to the present invention, the recorded audio level can be easily detected without decoding the audio data of the sub-band (SB), and reproduction is performed based on the detected audio level data. An object of the present invention is to provide an audio recording / reproducing apparatus and a recording medium in which the audio levels of a plurality of program tunes to be reproduced by controlling the audio output level are substantially the same.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating a configuration of an audio playback device according to an embodiment of the present invention.

FIG. 2 is a diagram illustrating an encoding / decoding unit of the audio recording / reproducing apparatus according to the first embodiment of the present invention.

FIG. 3 is a diagram illustrating an encoding / decoding unit of an audio recording / reproducing apparatus according to a second embodiment of the present invention.

FIG. 4 is a diagram illustrating audio data in which input audio data is divided into different SBs and bits are assigned in an audio recording / reproducing apparatus that records and reproduces audio by compressing and expanding the audio.

FIG. 5 is an MPE used in the audio recording / reproducing apparatus of the embodiment.
FIG. 3 is a schematic diagram illustrating an arrangement of various data in a G1 audio frame format.

FIG. 6 is a flowchart illustrating an operation of detecting an audio level in the audio recording / reproducing apparatus of the embodiment.

FIG. 7 is a flowchart illustrating an operation of automatically controlling the gain of the electronic volume at the time of reproduction in the audio recording / reproducing apparatus of the present embodiment.

FIG. 8 is a flowchart illustrating an operation of controlling the gain of the electronic volume in a case where the auxiliary data portion of the frame format signal is not used in another audio recording / reproducing apparatus of the embodiment.

[Explanation of symbols]

Reference Signs List 101 recording medium (IC memory), 102 interface unit 103 system microcomputer, 104 operation unit 105 display unit, 106 D /
F 107 D / A, 108 electronic volume 109 buffer circuit, 110 encoding / decoding unit 111 memory, 112 audio reproduction circuit unit 201 band division unit, 202 masking threshold calculation unit 203 bit allocation / scale factor calculation unit 204 requantization unit, 205 Formatting section 206 Level comparison / determination section 207 Scale factor maximum value detection section 207a Audio level detection section

Claims (4)

[Claims] 1. An audio recording / reproducing apparatus for compressing / expanding an audio signal to record / reproduce the audio signal, wherein the audio signal is divided into sub-bands of different audio frequency bands, quantized to obtain audio data, and a masking threshold is set. Masking threshold calculating means for calculating, level comparing means for comparing the level of the audio data with the masking threshold, bit allocation means for allocating bits to the sub-band based on the comparison result of the level comparing means, Requantization means for requantizing the audio data of the number of bits allocated to the subband by the allocation means, audio level detection means for detecting the audio level of the audio signal or audio data, and detection by the audio level detection means Data indicating the audio level is added to the auxiliary data section of the frame format. An audio recording / reproducing apparatus comprising: a signal conversion unit for converting a signal; and a control unit for controlling a gain of an audio level control unit based on data indicating the audio level when reproducing the signal of the frame format. 2. The audio recording / reproducing apparatus according to claim 1, wherein the bit allocation means calculates a scale factor of the sample data of the sub-band, and the audio level detecting section calculates a maximum value of the scale factor. Wherein the control means adds the maximum value of the scale factor to the auxiliary data section. 3. The audio recording / reproducing apparatus according to claim 1, wherein the audio level detecting unit detects an audio level of an audio signal or audio data input to the audio recording / reproducing apparatus, and the control unit includes: A voice recording / reproducing apparatus, wherein the voice level is added to the auxiliary data section. 4. A recording medium for recording a signal in a frame format in which data of a scale factor and audio data obtained by compressing audio data are arranged, wherein a maximum value of the scale factor or a value of the scale factor is stored in an auxiliary data portion of the frame format. A recording medium characterized by arranging and recording average values.