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TWI491179B - Encoding modulation system and method - Google Patents

Encoding modulation system and method Download PDF

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TWI491179B
TWI491179B TW098121271A TW98121271A TWI491179B TW I491179 B TWI491179 B TW I491179B TW 098121271 A TW098121271 A TW 098121271A TW 98121271 A TW98121271 A TW 98121271A TW I491179 B TWI491179 B TW I491179B
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bit
code
code stream
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TW201101707A (en
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Jen I Wu
Chen Wei Huang
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Hon Hai Prec Ind Co Ltd
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編碼調制系統及方法 Code modulation system and method

本發明涉及一種編碼技術,特別涉及一種編碼調制方法及使用該編碼調制方法的系統。 The present invention relates to an encoding technique, and more particularly to a coded modulation method and a system using the same.

ADPCM是一種自適應差分脈衝編碼調制方式,其透過適當比率對音頻信號進行壓縮,該編碼調制方式通常將當前採樣點採樣數據與前一採樣點採樣數據的差值編碼為4位元碼流,其中4位碼流包括1位元符號位元和3位元數值位。由於上述編碼調制方法中差值的符號位元需要占4位元碼流中的一位,僅有3位元用於編碼差值的數值位元,所以採用該種調制編碼方法的音頻信號保真性較低。 ADPCM is an adaptive differential pulse code modulation method that compresses an audio signal by an appropriate ratio. The code modulation method generally encodes the difference between the current sample point sample data and the previous sample point sample data into a 4-bit code stream. The 4-bit code stream includes a 1-bit symbol bit and a 3-bit value bit. Since the sign bit of the difference in the above coding and modulation method needs to occupy one bit in the 4-bit code stream, and only 3 bits are used to encode the value bit of the difference value, the audio signal using the modulation coding method is guaranteed. The truth is low.

有鑒於此,有必要提供一種保真性較好的編碼調制方法。 In view of this, it is necessary to provide a code modulation method with better fidelity.

另外,還有必要提供一種保真性較好的編碼調制系統。 In addition, it is also necessary to provide a code modulation system with better fidelity.

一種編碼調制方法,用於將音頻信號編碼為L位元碼流,包括以下步驟:採樣音頻信號的當前採樣點獲取採樣數據N,計算當前採樣點的採樣數據N與前一採樣點的採樣數據N-1的差值d(N); 判斷差值d(N)是否大於或者等於0;若差值d(N)大於或者等於0,則將差值d(N)的數值量化到M(2 L-1<M<2 L )個量化等級上,再根據量化等級將d(N)編碼為M個L位元數值碼流其中之一。 A code modulation method for encoding an audio signal into an L-bit code stream includes the steps of: sampling a current sample point of the audio signal to obtain sample data N, and calculating sample data N of the current sample point and sample data of the previous sample point The difference d(N) of N-1; whether the difference d(N) is greater than or equal to 0; if the difference d(N) is greater than or equal to 0, the value of the difference d(N) is quantized to M ( 2 L -1 < M <2 L ) Quantization levels, and d(N) is encoded into one of M L-bit value streams according to the quantization level.

一種編碼調制系統包括:用於將音頻信號編碼為L位元碼流,其包括:一採樣單元,用於採樣音頻信號各採樣點獲取各採樣點的採樣數據;一計算單元,計算當前採樣點的採樣數據N與前一採樣點的採樣數據N-1之間的差值d(N);一比較單元,比較差值d(N)是否大於或者等於0並輸出比較結果;其特徵在於:所述編碼調制系統還包括:一量化單元,用於將差值d(N)的數值量化到M(2 L-1<M<2 L )個量化等級上;一編碼單元,用於根據比較結果對差值d(N)進行編碼;若差值d(N)大於或者等於0,則編碼單元根據量化等級將d(N)編碼為M個L位元數值碼流其中之一。 A coding and modulation system includes: an audio signal for encoding an L-bit code stream, comprising: a sampling unit, configured to sample sampling data of each sampling point of each sampling point of the audio signal; and a calculating unit to calculate the current sampling point The difference d(N) between the sampled data N and the sampled data N-1 of the previous sample point; a comparison unit that compares whether the difference d(N) is greater than or equal to 0 and outputs a comparison result; The code modulation system further includes: a quantization unit for quantizing the value of the difference d(N) to M(2 L -1 < M <2 L ) quantization levels; a coding unit for comparing As a result, the difference d(N) is encoded; if the difference d(N) is greater than or equal to 0, the coding unit encodes d(N) into one of the M L-bit value streams according to the quantization level.

在上述編碼調制的方法及系統中,當前採樣點的採樣數據N與前一採樣點的採樣數據N-1的差值d(N)對應的數值和符號分別獨立編碼,使得數值碼流可以用L位表示,相對現有技術中數值碼流用L-1位表示,可以提高音頻信號的保真性。 In the above method and system for coding and modulation, the value and the symbol corresponding to the difference d(N) of the sampled data N of the current sampling point and the sampled data N-1 of the previous sampling point are separately encoded, so that the numerical code stream can be used. The L bit indicates that the fidelity of the audio signal can be improved compared to the prior art numerical code stream represented by the L-1 bit.

201‧‧‧採樣單元 201‧‧‧Sampling unit

203‧‧‧計算單元 203‧‧‧Computation unit

207‧‧‧量化單元 207‧‧‧Quantification unit

209‧‧‧編碼單元 209‧‧‧ coding unit

205‧‧‧比較單元 205‧‧‧Comparative unit

S101-S109‧‧‧編碼調制方法流程步驟 S101-S109‧‧‧ Code modulation method flow steps

圖1為一較佳實施方式的編碼調制方法流程圖。 1 is a flow chart of a code modulation method according to a preferred embodiment.

圖2為一較佳實施方式的編碼調制系統模組圖 2 is a block diagram of a code modulation system according to a preferred embodiment.

圖1為一較佳實施方式的編碼調制方法流程圖,其用於將音頻信號編碼為L位元碼流,所述編碼調制方法包括以下步驟。本實施方式以將音頻信號編碼為4位元碼流為例進行說明。 1 is a flow chart of a code modulation method according to a preferred embodiment for encoding an audio signal into an L-bit code stream, the code modulation method comprising the following steps. This embodiment will be described by taking an audio signal as a 4-bit code stream as an example.

步驟S101,採樣音頻信號的當前採樣點獲取採樣數據N。 In step S101, the current sampling point of the sampled audio signal acquires the sampled data N.

步驟S103,計算當前採樣點採樣數據N與前一採樣點採樣數據N-1的差值d(N)。 Step S103, calculating a difference d(N) between the current sample point sample data N and the previous sample point sample data N-1.

步驟S105,判斷差值d(N)是否大於或者等於0。若差值d(N)大於或者等於0,則進入步驟S107。若差值d(N)小於0,則進入步驟S109。 In step S105, it is determined whether the difference d(N) is greater than or equal to zero. If the difference d(N) is greater than or equal to 0, the process proceeds to step S107. If the difference d(N) is less than 0, the process proceeds to step S109.

步驟S107,將差值d(N)的數值量化到M(23<M<24)個量化等級上,再根據量化等級將d(N)編碼為M個4位元數值碼流其中之一。 Step S107, quantizing the value of the difference d(N) to M (2 3 < M < 2 4 ) quantization levels, and encoding d(N) into M 4-bit value streams according to the quantization level. One.

步驟S109,將差值d(N)的數值量化到M個量化等級上,並根據量化等級將d(N)編碼為M個4位元數值碼流其中之一後,再在該4位數值碼流前增加d(N)的符號對應的4位元符號碼流P。符號碼流P與M個數值碼流均不相同,其可以取24個L位碼流中另外24-M個4位碼流其中之一。在本實施方式中M可以為15。 Step S109, quantizing the value of the difference d(N) to M quantization levels, and encoding d(N) into one of the M 4-bit value streams according to the quantization level, and then in the 4-bit value. The 4-bit symbol code stream P corresponding to the sign of d(N) is added before the code stream. Symbol stream number P and the value of M streams are not the same, it may take two 24-bit code stream additionally L 24 - M 4-bit code stream one. In the present embodiment, M may be 15.

在上述編碼調制方法中,音頻信號的當前採樣點的採樣數據N與前一採樣點的採樣數據N-1的差值d(N)對應的數值和符號分別獨 立編碼,使得數值碼流可以用L位表示,相對現有技術中數值碼流用L-1位表示,可以提高音頻信號的保真性。 In the above code modulation method, the value and the symbol corresponding to the difference d(N) between the sampled data N of the current sample point of the audio signal and the sampled data N-1 of the previous sample point are respectively The vertical coding enables the numerical code stream to be represented by L bits. Compared with the prior art, the numerical code stream is represented by L-1 bits, which can improve the fidelity of the audio signal.

圖2為一較佳實施方式編碼調制系統模組圖,編碼調制系統200包括採樣單元201,計算單元203,比較單元205,量化單元207及編碼單元209。 2 is a block diagram of a coded modulation system according to a preferred embodiment. The code modulation system 200 includes a sampling unit 201, a calculation unit 203, a comparison unit 205, a quantization unit 207, and an encoding unit 209.

採樣單元201用於採樣音頻信號各採樣點獲取各採樣點的採樣數據以將連續的類比信號轉換為離散的數位信號。 The sampling unit 201 is configured to sample sampling points of the audio signal to obtain sampling data of each sampling point to convert the continuous analog signal into a discrete digital signal.

計算單元203用於計算當前採樣點的採樣數據N與前一採樣點的採樣數據N-1之間的差值d(N)。 The calculating unit 203 is configured to calculate a difference d(N) between the sampling data N of the current sampling point and the sampling data N-1 of the previous sampling point.

比較單元205用於比較差值d(N)是否大於或者等於0並輸出比較結果。 The comparison unit 205 is for comparing whether the difference d(N) is greater than or equal to 0 and outputting the comparison result.

量化單元207用於將差值d(N)的數值量化到M(23<M<24)個量化等級上。 The quantization unit 207 is for quantizing the value of the difference d(N) to M (2 3 < M < 2 4 ) quantization levels.

編碼單元209與比較單元205和量化單元207相連接,用於根據比較單元205的比較結果對量化單元207計算的差值d(N)進行編碼。若採樣數據N與前一採樣數據N-1的差值大於或者等於0,則根據量化等級將d(N)的數值編碼為M個4位元數值碼流其中之一。若採樣數據N與前一採樣數據N-1的差值小於0,則根據量化等級將d(N)的數值編碼為M個4位元數值碼流其中之一後,再在4位數值碼流前增加d(N)的符號對應的4位元符號碼流P。符號碼流P與M個數值碼流均不相同,其可以取24個4位碼流中另外24-M個4位碼流其中之一。在本實施方式中M可以為15。 The encoding unit 209 is connected to the comparison unit 205 and the quantization unit 207 for encoding the difference d(N) calculated by the quantization unit 207 according to the comparison result of the comparison unit 205. If the difference between the sampled data N and the previous sampled data N-1 is greater than or equal to 0, the value of d(N) is encoded as one of the M 4-bit valued code streams according to the quantization level. If the difference between the sampled data N and the previous sampled data N-1 is less than 0, the value of d(N) is encoded into one of the M 4-bit value streams according to the quantization level, and then the 4-digit value code is used. The 4-bit symbol code stream P corresponding to the sign of d(N) is added before the stream. Symbol stream number P and the value of M streams are not the same, it may take 24 4-bit code stream addition 24 - M 4-bit code stream one. In the present embodiment, M may be 15.

在上述編碼調制系統200中,當前採樣點的採樣數據N與前一採樣點的採樣數據N-1的差值d(N)對應的數值和符號分別獨立編碼,使得數值碼流可以用L位表示,相對現有技術中數值碼流用L-1位表示,可以提高音頻信號的保真性。 In the above code modulation system 200, the value and the symbol corresponding to the difference d(N) of the sampled data N of the current sampling point and the sampled data N-1 of the previous sampling point are independently encoded, so that the numerical code stream can use the L bit. It is indicated that the fidelity of the audio signal can be improved compared with the L-1 bit of the numerical code stream in the prior art.

綜上所述,本發明符合發明專利要件,爰依法提出專利申請。惟,以上所述者僅為本發明之較佳實施方式,本發明之範圍並不以上述實施方式為限,舉凡熟悉本案技藝之人士援依本發明之精神所作之等效修飾或變化,皆應涵蓋於以下申請專利範圍內。 In summary, the present invention complies with the requirements of the invention patent and submits a patent application according to law. However, the above description is only the preferred embodiment of the present invention, and the scope of the present invention is not limited to the above-described embodiments, and equivalent modifications or variations made by those skilled in the art in light of the spirit of the present invention are It should be covered by the following patent application.

S101-S109‧‧‧編碼調制方法流程步驟 S101-S109‧‧‧ Code modulation method flow steps

Claims (6)

一種編碼調制方法,用於將音頻信號編碼為L位元碼流,包括如下步驟:採樣音頻信號的當前採樣點獲取採樣數據N;計算當前採樣點的採樣數據N與前一採樣點的採樣數據N-1的差值d(N);判斷差值d(N)是否大於或者等於0;若差值d(N)大於或者等於0,則將差值d(N)的數值量化到M(2 L-1<M<2 L )個量化等級上,再根據量化等級將d(N)編碼為M個L位元數值碼流其中之一;若d(N)小於0,則將差值d(N)的數值量化到M(2 L-1<M<2 L )個量化等級上,並根據量化等級將d(N)編碼為M個L位元數值碼流其中之一後,再在該L位數值碼流前增加d(N)的符號對應的L位元符號碼流P,其中符號碼流P與M個L數值碼流均不相同。 A code modulation method for encoding an audio signal into an L-bit code stream includes the steps of: sampling a current sample point of the audio signal to obtain sample data N; calculating sample data N of the current sample point and sample data of the previous sample point The difference d(N) of N-1; whether the difference d(N) is greater than or equal to 0; if the difference d(N) is greater than or equal to 0, the value of the difference d(N) is quantized to M ( 2 L -1 < M <2 L ) quantization levels, and d(N) is encoded into one of M L-bit value streams according to the quantization level; if d(N) is less than 0, the difference is The value of d(N) is quantized to M(2 L -1 < M <2 L ) quantization levels, and d(N) is encoded into one of M L-bit value streams according to the quantization level, and then An L-bit symbol code stream P corresponding to the symbol of d(N) is added before the L-bit value code stream, wherein the symbol code stream P and the M L-value code streams are different. 如申請專利範圍第1項所述之編碼調制方法,其中該符號碼流P取2 L 個L位元碼流中另外2 L -M個L位元碼流其中之一。 If the application code modulation method according to item 1 of the scope of the patent, wherein stream number P takes the symbol L 2 L-bit code stream additional 2 L - M L-stream one byte code. 如申請專利範圍第1項所述之編碼調制方法,其中該L為4,該M為15。 The code modulation method according to claim 1, wherein the L is 4 and the M is 15. 一種編碼調制系統,用於將音頻信號編碼為L位元碼流,其包括:一採樣單元,用於採樣音頻信號各採樣點獲取各採樣點的採樣數據;一計算單元,計算當前採樣點的採樣數據N與前一採樣點的採樣數據N-1之間的差值d(N);一比較單元,比較差值d(N)是否大於或者等於0並輸出比較結果;其特徵在於:所述編碼調制系統還包括:一量化單元,用於將差值d(N)的數值量化到M(2 L-1<M<2 L )個量 化等級上;一編碼單元,用於根據比較結果對差值d(N)進行編碼;若差值d(N)大於或者等於0,則編碼單元根據量化等級將d(N)編碼為M個L位元數值碼流其中之一;d(N)若小於0,則編碼單元根據量化等級將d(N)編碼為M個L位元數值碼流其中之一後,再在該L位數值碼流前增加d(N)的符號對應的L位元符號碼流P,且符號碼流P與M個L為數值碼流均不相同。 A code modulation system for encoding an audio signal into an L-bit code stream, comprising: a sampling unit for sampling sampling points of an audio signal to obtain sampling data of each sampling point; and a calculating unit for calculating a current sampling point a difference d(N) between the sampled data N and the sampled data N-1 of the previous sample point; a comparison unit that compares whether the difference d(N) is greater than or equal to 0 and outputs a comparison result; The code modulation system further includes: a quantization unit for quantizing the value of the difference d(N) to M(2 L -1 < M <2 L ) quantization levels; a coding unit for comparing results Encoding the difference d(N); if the difference d(N) is greater than or equal to 0, the coding unit encodes d(N) into one of the M L-bit value streams according to the quantization level; d(N) If it is less than 0, the coding unit encodes d(N) into one of the M L-bit value code streams according to the quantization level, and then adds L corresponding to the symbol of d(N) before the L-bit value code stream. The bit symbol code stream P, and the symbol code stream P and the M L numbers are different from each other. 如申請專利範圍第4項所述之編碼調制系統,其中該符號碼流P取2 L 個L位碼流中另外2 L -M個L位碼流其中之一。 If the application code modulation system as recited in item 4 patentable scope, wherein the symbol code stream P 2 taken L L-bit code stream additional 2 L - M L-bit code stream one. 如申請專利範圍第4項所述之編碼調制系統,其中該L為4,該M為15。 The code modulation system of claim 4, wherein the L is 4 and the M is 15.
TW098121271A 2009-06-24 2009-06-24 Encoding modulation system and method TWI491179B (en)

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Citations (4)

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Publication number Priority date Publication date Assignee Title
US3689840A (en) * 1971-04-29 1972-09-05 Bell Telephone Labor Inc Coding of sign information in dpcm systems
US6122618A (en) * 1997-04-02 2000-09-19 Samsung Electronics Co., Ltd. Scalable audio coding/decoding method and apparatus
US6625574B1 (en) * 1999-09-17 2003-09-23 Matsushita Electric Industrial., Ltd. Method and apparatus for sub-band coding and decoding
TWI226035B (en) * 2003-10-16 2005-01-01 Elan Microelectronics Corp Method and system improving step adaptation of ADPCM voice coding

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US3689840A (en) * 1971-04-29 1972-09-05 Bell Telephone Labor Inc Coding of sign information in dpcm systems
US6122618A (en) * 1997-04-02 2000-09-19 Samsung Electronics Co., Ltd. Scalable audio coding/decoding method and apparatus
US6625574B1 (en) * 1999-09-17 2003-09-23 Matsushita Electric Industrial., Ltd. Method and apparatus for sub-band coding and decoding
TWI226035B (en) * 2003-10-16 2005-01-01 Elan Microelectronics Corp Method and system improving step adaptation of ADPCM voice coding

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