JPH0366225A - Method for compressing music signal - Google Patents
Method for compressing music signalInfo
- Publication number
- JPH0366225A JPH0366225A JP1203390A JP20339089A JPH0366225A JP H0366225 A JPH0366225 A JP H0366225A JP 1203390 A JP1203390 A JP 1203390A JP 20339089 A JP20339089 A JP 20339089A JP H0366225 A JPH0366225 A JP H0366225A
- Authority
- JP
- Japan
- Prior art keywords
- samples
- music signal
- encoded
- sample
- linear approximation
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
Links
- 238000000034 method Methods 0.000 title claims abstract description 17
- 230000006835 compression Effects 0.000 claims description 13
- 238000007906 compression Methods 0.000 claims description 13
- 238000006243 chemical reaction Methods 0.000 abstract 2
- 230000005540 biological transmission Effects 0.000 abstract 1
- 239000000523 sample Substances 0.000 description 20
- 238000010586 diagram Methods 0.000 description 7
- 230000006866 deterioration Effects 0.000 description 3
- 230000000694 effects Effects 0.000 description 3
- 238000013139 quantization Methods 0.000 description 2
- 230000005236 sound signal Effects 0.000 description 2
Landscapes
- Electrophonic Musical Instruments (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
Abstract
Description
【発明の詳細な説明】
産業上の利用分野
本発明は、Ａ／Ｄ変換された音声信号のサンプル系列を
直線近似する音楽信号圧縮方法に関するものである。DETAILED DESCRIPTION OF THE INVENTION Field of the Invention The present invention relates to a music signal compression method that linearly approximates a sample sequence of an A/D converted audio signal.
従来の技術
従来の音声符号化方式においては、音声信号は複数の帯
域に分割され、各々について予測係数が求められ、この
予測係数を用いて、各々の帯域の信号は予測符号化され
る。その際に量子化レベル数を時間的に不均一に割り当
て、符号化前後の量子化誤差が小さくなるようにする（
例えば、特願昭５４−４２８５８号参照）。2. Description of the Related Art In conventional audio encoding systems, an audio signal is divided into a plurality of bands, prediction coefficients are determined for each band, and the signals in each band are predictively encoded using the prediction coefficients. At that time, the number of quantization levels is allocated temporally non-uniformly so that the quantization error before and after encoding becomes small (
For example, see Japanese Patent Application No. 54-42858).
発明が解決しようとする課題
従来の音声符号化方式では、分割された各周波数帯域に
ついて予測残差信号を送っており、この予測残差信号の
情報量が比較的多く、全体としての伝送情報量が多くな
るという問題点があった。Problems to be Solved by the Invention In conventional audio encoding systems, a prediction residual signal is sent for each divided frequency band, and the amount of information in this prediction residual signal is relatively large. There was a problem that there were many.
本発明は、少ない伝送情報量で品質がよい音楽信号圧縮
方法を提供することを目的としている。An object of the present invention is to provide a music signal compression method that achieves high quality with a small amount of transmitted information.
課題を解決するための手段
上記課題を解決するために、本発明の音楽信号圧縮方法
は、Ａ／Ｄ変換された音楽信号のサンプル系列を一定数
のサンプルを含むブロック毎に近似するもので、上記ブ
ロック内の最初の３サンプルを近似する直線を最小２乗
法を用いて求め、最１Ｊ）２乗誤差が上記ブロックに含
まれる全サンプルの振幅の２乗和に基づいてあらかじめ
定めておいたしきい値を超えていなければ、順にサンプ
ル数を１個ずつ増やしながら、逐次最小２乗誤差を算出
し、最小２乗誤差がしきい値を超えたとすれば、しきい
値を超えるまでのサンプルを近似する直線を定める係数
２個と、直線近似されたサンプル個数を符号化すること
とし、次の３サンプルから新たな直線近似を行なうよう
にする。最初の３サンプルの直線近似の最小２乗誤差が
しきい値を超えてしまった場合は、最初の２サンプルの
値を２サンプルが直線近似されていないことのフラグを
符号化することとし、３個目のサンプルから新たな直線
近似を行なうようにする。Means for Solving the Problems In order to solve the above problems, the music signal compression method of the present invention approximates a sample sequence of an A/D converted music signal for each block containing a certain number of samples. A straight line approximating the first three samples in the above block is found using the least squares method, and the maximum 1 J) squared error is a predetermined threshold based on the sum of squares of the amplitudes of all samples included in the above block. If the value is not exceeded, the least squares error is calculated successively by increasing the number of samples one by one, and if the least squares error exceeds the threshold, approximate the samples until the threshold is exceeded. The two coefficients that define the straight line and the number of samples subjected to linear approximation are encoded, and a new linear approximation is performed from the next three samples. If the least squares error of the linear approximation of the first 3 samples exceeds the threshold, the values of the first 2 samples will be encoded as a flag indicating that the 2 samples are not linearly approximated. A new straight line approximation is performed starting from the second sample.
また、直線近似されないで符号化されるサンプルがある
とき、そのサンプルを対数圧縮または下位ｎビット切り
捨てて符号化することもできる。Furthermore, when there is a sample to be encoded without linear approximation, the sample can be encoded by logarithmic compression or by truncating the lower n bits.
作用
上記のような音楽信号圧縮方法を用いてサンプル系列を
符号化すると、最小２乗誤差があらかじめ定めたしきい
値を超えないサンプル系列が一つの直線として扱われ、
サンプル系列は圧縮されて符号されることとなる。また
、３サンプルですでにしきい値を超えるような、サンプ
ルの値の変化が激しい場合はサンプルの値をそのまま符
号化することにより、音質の劣化が少なくなる。Effect: When a sample sequence is encoded using the music signal compression method described above, the sample sequence whose least squares error does not exceed a predetermined threshold is treated as a straight line.
The sample sequence will be compressed and encoded. Furthermore, when the sample values change rapidly, such as when the threshold value is already exceeded after three samples, the deterioration of sound quality is reduced by encoding the sample values as they are.
実施例
以下本発明の音楽信号圧縮方法の一実施例について、図
面を参照しながら説明する。第１図は音楽信号の模式図
である。第１図において１は音楽信号を表わし、ｓｌ、
ｓ２、ｓ３、ｓ４、ｓ ｓ ”’ ”’はＡ／Ｄ変換さ
れた音楽信号のサンプル系列である。２．３．４．５は
音楽信号を近似する直線である。Embodiment Hereinafter, an embodiment of the music signal compression method of the present invention will be described with reference to the drawings. FIG. 1 is a schematic diagram of a music signal. In FIG. 1, 1 represents a music signal, sl,
s2, s3, s4, s s ``'''''' are sample sequences of A/D converted music signals. 2.3.4.5 is a straight line that approximates the music signal.
ｓＩ’ 、”２’ 、Ｓ３”、５４’ 、ｓｓ′ ・
・・・・・はサンプルＳ０、ｓ２、ｓ３、ｓ４、Ｓ、・
・・・・・の近似値である。直線２．３．４．５は最小
２乗誤差があらかじめ定めたしきい値を超えないような
直線として求められる。sI', "2', S3", 54', ss'・
...are samples S0, s2, s3, s4, S,...
It is an approximate value of... Straight line 2.3.4.5 is determined as a straight line such that the least square error does not exceed a predetermined threshold.
第２図は第１図の直線の係数２個、サンプル個数を符号
化した図である。６．７．８．９は各々第１図の直線２
．３．４．５を符号化したもで、ａｌ、ａｌ、ａ３、ａ
４およびｔｅｌｓ ｂ２、ｂ３、ｂ４は直線をさだめる
係数、ｎｌ、ｎ２、ｎ３、ｎ４はサンプル個数である。FIG. 2 is a diagram in which two coefficients of the straight line in FIG. 1 and the number of samples are encoded. 6.7.8.9 are the straight lines 2 in Fig. 1, respectively.
．． 3.4.5 is encoded, al, al, a3, a
4 and tels b2, b3, and b4 are coefficients for forming a straight line, and nl, n2, n3, and n4 are the numbers of samples.
第１図において１サンプル当り１６ビツトで符号化され
ているとすると、図中には１８サンプルあるので、全サ
ンプルで２８８ビツト必要である。しかし、本方式を用
いて、第２図のように符号化した場合、直線の係数２個
、サンプル個数を各々８ビツトで符号化すると、４直線
あるので、全サンプルで９６ビツト必要となり、情報圧
縮が可能となる。Assuming that each sample is encoded with 16 bits in FIG. 1, there are 18 samples in the figure, so 288 bits are required for all samples. However, when encoding as shown in Figure 2 using this method, if the two straight line coefficients and the number of samples are encoded with 8 bits each, there are 4 straight lines, so 96 bits are required for all samples, and the information Compression becomes possible.
第３図は第１図と同様に音楽信号の模式図である。第３
図において１０は音楽信号を表わし、Ｓｌ、ｓ２、ｓ３
、ｓ４、Ｓ３、Ｓ ６ ”’ ”’はＡ／Ｄ変換された
音楽信号のサンプル系列である。１１．１２．１３．１
４は音楽信号を近似する直線である。直線１１．１２．
１３．１４は最小２乗誤差があらかじめ定めたしきい値
を超えないような直線として求められる。Similar to FIG. 1, FIG. 3 is a schematic diagram of a music signal. Third
In the figure, 10 represents music signals, Sl, s2, s3
, s4, S3, S 6 ``'''''' are sample sequences of A/D converted music signals. 11.12.13.1
4 is a straight line that approximates the music signal. Straight line 11.12.
13.14 is determined as a straight line such that the least square error does not exceed a predetermined threshold.
第４図は第３図の直線の係数２個、サンプル個数を符号
化した図である。１５．１６．１７は各々第３図のサン
プルＳ１、ｓ２、とフラグＦ１、サンプルＳ３、Ｓ、と
フラグＦ２、サンプルＳ９、ｓ６とフラグＦ３を符号化
したものである。１８．１９．２０．２１は各々第３図
の直線１１．１２．１３．１４を符号化したもので、ａ
、、ａ２、ａ３、ａ４およびｂｌ、ｂ２、ｂ３、ｂ４は
直線を定める係数、ｎｌ、ｎ２、ｎ３、ｎ４はサンプル
個数である。FIG. 4 is a diagram in which two coefficients and the number of samples of the straight line in FIG. 3 are encoded. 15, 16, and 17 are encoded samples S1, s2, flag F1, samples S3, S, flag F2, samples S9, s6, and flag F3 in FIG. 3, respectively. 18, 19, 20, and 21 are the encoded lines 11, 12, 13, and 14 in Figure 3, respectively, and a
, , a2, a3, a4, bl, b2, b3, and b4 are coefficients that define straight lines, and nl, n2, n3, and n4 are the numbers of samples.
第３図においてエサンプル当り１６ビツトで符号化され
ているとすると、図中には１８サンプルあるので、全サ
ンプル２８８ビツト必要である。しかし、本方式を用い
て、第４図に示すように符号化した場合、直線の係数２
個、サンプル個数、フラグを各々８ビツトで符号化する
と、全サンプルで２１６ビツト必要となり、情報圧縮が
可能となる。Assuming that each sample is encoded with 16 bits in FIG. 3, there are 18 samples in the figure, so 288 bits are required for all samples. However, when using this method and encoding as shown in Figure 4, the linear coefficient 2
If each sample, number of samples, and flag are encoded with 8 bits, 216 bits will be required for all samples, making it possible to compress information.
発明の効果
本発明は、以上説明したような音楽信号圧縮方法である
ので、以下に記載されるような効果を奏する。Effects of the Invention Since the present invention is a music signal compression method as described above, it has the following effects.
すなわち、本発明はあらかじめＡ／Ｄ変換された音楽信
号のサンプル系列を直線近似するので、情報伝送量が少
なくなる。また、最小２乗法を用いて直線を求めるので
、サンプル系列をよりよく近似する直線が得られる。さ
らに、３サンプルを近似する直線を求めた際に、最小２
乗誤差があらかじめ定めたしきい値を超える場合は、最
初の２サンプルをそのまま符号化するので、サンプルの
値の変化が急激であっても、音質の劣化が少なくなる。That is, since the present invention linearly approximates a sample sequence of a music signal that has been A/D converted in advance, the amount of information transmitted is reduced. Furthermore, since the straight line is determined using the least squares method, a straight line that better approximates the sample series can be obtained. Furthermore, when finding a straight line that approximates 3 samples, the minimum 2
If the multiplicative error exceeds a predetermined threshold, the first two samples are encoded as they are, so even if the sample values change rapidly, there is little deterioration in sound quality.
また、ブロック中に含まれるサンプルの振幅に合わせて
ブロック毎にしきい値を定めるので、よりよい音質を確
保することができる。さらに、ブロック毎に符号化する
ことにより、他の信号と同期をとることが容易となる。Furthermore, since the threshold value is determined for each block according to the amplitude of the samples included in the block, better sound quality can be ensured. Furthermore, by encoding each block, it becomes easy to synchronize with other signals.
また、サンプル値を直線近似しないでそのまま符号化す
る際に、対数圧縮または下位ｎビット切り捨てて符号化
することもできるので、音質の劣化を押さえて、より情
報伝送量を少なくすることが可能である。Furthermore, when encoding sample values as they are without linear approximation, it is also possible to perform logarithmic compression or truncate the lower n bits before encoding, which reduces the amount of information transmitted while minimizing deterioration in sound quality. be.
第１図は本発明の一実施例を説明するための音楽信号の
模式図、第２図は第１図の音楽信号を符号化した説明図
、第３図は本発明の一実施例を説明するための音楽信号
の模式図ぎ、第４図は第３図の音楽信号を符号化した説
明図である。
１・・・・・・音楽信号、２．３．４．５・・・・・・
音楽信号を近似する直線、６．７．８．９・・・・・・
符号化された音楽信号、１０・・・・・・音楽信号、１
１１２．１３．１４・・・・・・音楽信号を近似する直
線、１５．１６．１７．１８．１９．２０．２１・・・
・・・符号化された音楽信号。Fig. 1 is a schematic diagram of a music signal for explaining an embodiment of the present invention, Fig. 2 is an explanatory diagram of the encoded music signal of Fig. 1, and Fig. 3 is an illustration of an embodiment of the present invention. FIG. 4 is an explanatory diagram of the encoded music signal of FIG. 3. 1...Music signal, 2.3.4.5...
Straight line approximating the music signal, 6.7.8.9...
Encoded music signal, 10...Music signal, 1
112.13.14... Straight line approximating the music signal, 15.16.17.18.19.20.21...
...Encoded music signal.
Claims (3)
系列を一定数のサンプルを含むブロック毎に直線近似す
る音楽信号圧縮方法であって、上記ブロック内の各一定
時刻における最初の３サンプルを近似する直線を最小２
乗法を用いて求め、最小２乗誤差が、上記ブロックに含
まれるサンプルの振幅の２乗和に基づいてあらかじめ定
めておいたしきい値を超えていなければ、順にサンプル
数を１個ずつ増やしながら、逐次最小２乗誤差を算出し
、上記最小２乗誤差が上記しきい値を超えたとすれば、
サンプルを近似する直線を定める係数２個と、直線近似
されたサンプル数ｎ（ｎは整数）を符号化して次の第（
ｎ＋１）サンプルから新たな直線近似を行なうようにし
、前記最初の３サンプルの直線近似の最小２乗誤差がす
でに上記しきい値を超えた場合は、最初の２サンプルの
値と前記最初の２サンプルが直線近似されていないこと
のフラグを符号化して第３サンプルから新たな直線近似
を行なうようにし、ブロック内のすべてのサンプルの直
線近似を完了したならば、次のブロックから新たに直線
近似を開始し、Ａ／Ｄ変換された音楽信号のサンプル系
列すべてを直線近似するようにしたことを特徴とする音
楽信号圧縮方法。(1) A music signal compression method in which a sample sequence of a music signal that has been A/D converted in advance is linearly approximated for each block containing a certain number of samples, and the first three samples at each certain time in the block are approximated. Minimum of 2 straight lines
If the least square error does not exceed a predetermined threshold based on the sum of the squares of the amplitudes of the samples included in the block, the number of samples is increased one by one, and If the successive least squares error is calculated and the above least squares error exceeds the above threshold,
The two coefficients that define the straight line that approximates the samples and the number of linearly approximated samples n (n is an integer) are encoded and the following (
n+1) If a new linear approximation is performed from the samples, and the least square error of the linear approximation of the first three samples already exceeds the above threshold, the values of the first two samples and the first two samples are encodes a flag indicating that is not linearly approximated and performs a new linear approximation from the third sample, and when the linear approximation of all samples in the block is completed, a new linear approximation is performed from the next block. A music signal compression method characterized in that all sample sequences of music signals that have been started and A/D converted are linearly approximated.
Ｓｉ＋１（ｉは整数）があるとき、上記Ｓｉ、Ｓｉ＋１
を対数圧縮して符号化するようにした請求項（１）記載
の音楽信号圧縮方法。(2) Sample Si encoded without linear approximation,
When Si+1 (i is an integer), the above Si, Si+1
2. The music signal compression method according to claim 1, wherein the music signal is encoded by logarithmically compressing it.
Ｓｉ＋１（ｉは整数）があるとき、上記Ｓｉ、Ｓｉ＋１
を下位ｎビット切捨てを行なって符号化するようにした
請求項（１）記載の音楽信号圧縮方法。(3) Sample Si encoded without linear approximation,
When Si+1 (i is an integer), the above Si, Si+1
2. The music signal compression method according to claim 1, wherein the music signal is encoded by truncating the lower n bits.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP1203390A JPH0787385B2 (en) | 1989-08-05 | 1989-08-05 | Music signal compression method |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP1203390A JPH0787385B2 (en) | 1989-08-05 | 1989-08-05 | Music signal compression method |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| JPH0366225A true JPH0366225A (en) | 1991-03-20 |
| JPH0787385B2 JPH0787385B2 (en) | 1995-09-20 |
Family
ID=16473253
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| JP1203390A Expired - Lifetime JPH0787385B2 (en) | 1989-08-05 | 1989-08-05 | Music signal compression method |
Country Status (1)
| Country | Link |
|---|---|
| JP (1) | JPH0787385B2 (en) |
Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP2007031903A (en) * | 2005-07-28 | 2007-02-08 | Kengo Kojima | Five-toe socks |
| CN100435136C (en) * | 2006-06-21 | 2008-11-19 | 浙江中控软件技术有限公司 | Real-time data compression method based on least square linear fit |
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| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JPS5352045A (en) * | 1976-10-22 | 1978-05-12 | Fujitsu Ltd | Modulating converter for pcm-forecasting code |
| JPS53101968A (en) * | 1977-02-17 | 1978-09-05 | Fujitsu Ltd | Modulating converter for pcm-forecast code |
| JPS5996513A (en) * | 1982-11-24 | 1984-06-04 | Nippon Gakki Seizo Kk | Method for recording and reproducing waveform |
| JPS63217731A (en) * | 1987-03-05 | 1988-09-09 | Yuuseishiyou Tsushin Sogo Kenkyusho | Method for saving memory quantity for vector quantized code directory |
| JPS6482711A (en) * | 1987-09-25 | 1989-03-28 | Hitachi Ltd | Encoding device |
| JPH01105617A (en) * | 1987-10-19 | 1989-04-24 | Sanyo Electric Co Ltd | Digital correction circuit |
| JPH02205117A (en) * | 1989-02-03 | 1990-08-15 | Matsushita Electric Ind Co Ltd | Music signal compression method |
-
1989
- 1989-08-05 JP JP1203390A patent/JPH0787385B2/en not_active Expired - Lifetime
Patent Citations (7)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JPS5352045A (en) * | 1976-10-22 | 1978-05-12 | Fujitsu Ltd | Modulating converter for pcm-forecasting code |
| JPS53101968A (en) * | 1977-02-17 | 1978-09-05 | Fujitsu Ltd | Modulating converter for pcm-forecast code |
| JPS5996513A (en) * | 1982-11-24 | 1984-06-04 | Nippon Gakki Seizo Kk | Method for recording and reproducing waveform |
| JPS63217731A (en) * | 1987-03-05 | 1988-09-09 | Yuuseishiyou Tsushin Sogo Kenkyusho | Method for saving memory quantity for vector quantized code directory |
| JPS6482711A (en) * | 1987-09-25 | 1989-03-28 | Hitachi Ltd | Encoding device |
| JPH01105617A (en) * | 1987-10-19 | 1989-04-24 | Sanyo Electric Co Ltd | Digital correction circuit |
| JPH02205117A (en) * | 1989-02-03 | 1990-08-15 | Matsushita Electric Ind Co Ltd | Music signal compression method |
Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP2007031903A (en) * | 2005-07-28 | 2007-02-08 | Kengo Kojima | Five-toe socks |
| CN100435136C (en) * | 2006-06-21 | 2008-11-19 | 浙江中控软件技术有限公司 | Real-time data compression method based on least square linear fit |
Also Published As
| Publication number | Publication date |
|---|---|
| JPH0787385B2 (en) | 1995-09-20 |
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