TWM438765U - Audio quantization and de-quantization device - Google Patents
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M438765 貧料(語音訊號的正、負值)。所以第一語音資料經過減碼轉換器22〇減碼之 後,得到新的7個第一語音資料為1i111〇11、彳111〇〇11、彳彳彳彳彳彳糾、 00000010、00000101、00001000、〇〇〇〇1111( 5、·13、小 2、5、8、15)。 接著’量化器230將第-語音資料量化而產生數位碼,而量化的方式 可利用查表法來進行。雖然語音資料有正負之別,但為了節省記憶體的使用, 通* /、會建立正半週的量化表(qUantizati〇n taWe)。在進行量化程序之前, 先將代表語音資料隸屬於正半週或負半週的符號位元紀錄下來, 然後再將 _ 4音資料取絕龍,侧正半週的量絲將語音資料量化。以下列舉一個 5bit表格,將第一語音資料(_5、_13、q、2、5、8、15)利用量化表1進 行量化程序Μ列如:紀錄語音資料_5、·13、μ的符號位元為】而2、5、8、 15的付號位元為〇,再將所有資料取絕對值之後得到(5、a、1、2、58、 15),其中5根據量化表1制最佳的索引碼為3,而對應的量化數位碼為 00011, 13根據量化表】得到最佳的索引碼為7,而對應的量化數位碼為 00111,在第五位元加上符號位元之後的數位資料分別是1〇〇11,1〇111。 _ 於疋’第-語音資料(-5、-13、-1、2、5、8、15)的絕對值根據表1得 到索引碼為(3、7、1、2、3、4、7),其對應之二進位數位瑪為(0001 ]、00111、 00001、00010、0001Ί、〇〇1〇〇、〇〇111),在第5位元替換上符號位元之 後的一進位數位碼為(loon、mm、10001、00010、〇〇〇11、〇 ⑴ 〇〇、 00111)。其中’ 13所對映的表格瑪,最接近者為15,因此,選擇其為對應 的表值。 表1 5 數位碼 索引碼 量化表格碼 00000 0 0 00001 1 Ί - " 00010 2 2 00011 3 "5 ' 00100 4 8 00101 5 ~9 00110 6 10 ' 00111 7 15 ' 01000 8 30 01001 9 "40 " 01010 10 "45 " 01011 「11 50 01100 12 ~55 " ' 01101 13 卜60 01110 14 90 01111 15 1〇〇 ' ' 11111 31 訊框切換控制碼 M438765 在實際的應用上,為了降低量化誤差,經常會使用多個量化表來對應 不同動態範圍的語音資料。請參考第4圖,數位碼6〇〇由符號資料612再 加上數字資料614構成。串接語音資料為多個數位碼6〇〇所組成而7筆數 位碼600的資料量共有35 bit。串接語音資料通常亦包含訊框標記資料 6〇6(Frame Header) ’訊框標記資料606記載當前的訊框所對應最佳量化 表的索引,以及訊框切換控制碼(一般是採用和語音編碼資料相同位元數但 不重複的特殊碼)。訊框標記資料606的資料長度端視量化表的個數而定, 例如’當採用8個量化表時訊框標記資料606需要3bit來對應最佳量化表 的索引,採用32個量化表時訊框標記資料606需要5bit來對應最佳量化表 的索引。以10 bit訊框標記資料長度為例(5bit訊框切換控制碼加上5bit最 佳量化表的索引),此串接語音資料經過编碼後的大小為35+10=45bit。於 M438765 M438765M438765 Poor material (positive and negative values of voice signals). Therefore, after the first voice data is reduced by the down code converter 22, the new seven first voice data are obtained as 1i111〇11, 彳111〇〇11, 彳彳彳彳彳彳 、, 00000010, 00000101, 00001000, 〇〇〇〇1111 (5, ·13, small 2, 5, 8, 15). Next, the quantizer 230 quantizes the first speech data to generate a digital code, and the manner of quantization can be performed by a look-up table method. Although the voice data is positive or negative, in order to save the use of memory, a positive half-week quantization table (qUantizati〇n taWe) will be established. Before performing the quantification process, the symbolic bits representing the speech data belonging to the positive half cycle or the negative half cycle are recorded, and then the _ 4 tone data is taken to the absolute, and the side positive half cycle is used to quantize the speech data. The following lists a 5-bit table, and uses the quantization table 1 to quantize the first speech data (_5, _13, q, 2, 5, 8, 15), such as: record the speech data _5, · 13, μ symbolic bits The yuan is] and the 2, 5, 8, and 15 paying bits are 〇, and then all the data are taken as absolute values (5, a, 1, 2, 58, 15), 5 of which are based on the quantization table 1 The preferred index code is 3, and the corresponding quantized digit code is 00011, 13 according to the quantization table, the best index code is 7, and the corresponding quantized digit code is 00111, after the fifth bit is added with the sign bit. The digital data is 1〇〇11,1〇111. _ Yu's absolute value of the first-speech data (-5, -13, -1, 2, 5, 8, 15) according to Table 1 to get the index code (3, 7, 1, 2, 3, 4, 7) ), the corresponding binary digits are (0001), 00111, 00001, 00010, 0001Ί, 〇〇1〇〇, 〇〇111), and the digit of the digit after the fifth digit is replaced by the sign bit is (loon, mm, 10001, 00010, 〇〇〇11, 〇(1) 〇〇, 00111). Among the 13 tables, the closest one is 15, so select it as the corresponding table value. Table 1 5 Digital Code Index Code Quantization Table Code 00000 0 0 00001 1 Ί - " 00010 2 2 00011 3 "5 ' 00100 4 8 00101 5 ~9 00110 6 10 ' 00111 7 15 ' 01000 8 30 01001 9 " 40 " 01010 10 "45 " 01011 "11 50 01100 12 ~55 " ' 01101 13 Bu 60 01110 14 90 01111 15 1〇〇' ' 11111 31 Frame switching control code M438765 In practical applications, To reduce the quantization error, multiple quantization tables are often used to correspond to different dynamic range speech data. Please refer to Figure 4, the digit code 6〇〇 is composed of symbol data 612 plus digital data 614. The concatenated speech data is multiple The digital code consists of 6 而 and the 7 digits of the code 600 has a total of 35 bits. The serial voice data usually also contains the frame marker data 6 〇 6 (Frame Header) 'frame marker data 606 records the current frame The index corresponding to the optimal quantization table, and the frame switching control code (generally a special code that uses the same number of bits as the voice encoded data but does not repeat). The data length of the frame mark data 606 depends on the number of quantization tables. Example 'When 8 quantization tables are used, the frame mark data 606 needs 3 bits to correspond to the index of the best quantization table, and 32 quantization table time frame mark data 606 needs 5 bits to correspond to the index of the best quantization table. The length of the frame tag data is an example (the index of the 5-bit frame switching control code plus the 5-bit optimal quantization table), and the size of the serialized speech data is 35+10=45 bits. On M438765 M438765
五.、新型說明: 【新型所屬之技術領域】 本創作係為一種量化裝置,特別是關於一種音訊量化裝置, 【先前技術】 語音信號原為類比信號,經過數位化及壓縮會產生失真,一般而言壓 縮率較高,信號失真較大,但所需傳輸碼率較低。所以在傳輸頻寬不足情 況下’在可韻通話内容的餅下’通常會選擇驗輪高的肢。如果 沒有傳輸頻寬的問題,一般採用信號失真較小G711協定是較好的選擇。 請參考第1圖,其為先前技術之語音編瑪與解碼系統圖,包含:語音 輸入訊號100、語音編碼器200、記憶體300、語音解碼器400、語音輸出 訊號500。其中,語音輸入訊號鄕為—段真實的聲音,其為類比訊號。 舉例而言,語音編碼器200若為16 bit的單聲道,若以每秒8KHz的頻率 取樣’資料量為每秒12_卜當語音輸入訊號1〇〇輸入至語音編碼器2〇〇, 語音輸入峨即會被取樣麟秒128kbit料聲道麵,秘過塵縮 編碼後’儲存在記憶體訓+。語音編碼器2㈤在實際上的應肖,即為一 種壓縮器。於實際的應用上,树為了降低記憶體3㈤的使用量,一般會 把16邱❾語音資料壓縮為較低的解析度資料(如5bit或4bit)並存在記憶體 内’即可有效降低記憶體300的使用量。最後,語音解碼器4〇〇會將 記憶體300崎儲存壓縮過後的較低的解析度資料解讀,再轉換成具有 1哪的單聲道語音資料’並轉換為語音輸出訊號500。 接著’清參考第2A圖’係為先前技術的語音編碼器2〇〇之詳細方塊圖。 其中,語音編石馬器200包含:類比數位轉換器21〇、減碼轉換器22〇、量 3 M438765 I柄月f日雙2 化器230與資料編· 24Ge其中,數位轉換器加接收類 輸入訊號而轉換為數位的第一語音資料。減碼轉換$ 22〇連接類比數位轉 換器210 ’對第-語音資料進行減碼。量化器23〇連接減碼轉換器挪, 接枚第-語音資料並進行量化而產生一數位碼,該數位碼包含符號資料與- 數字資料。資料編碼器240連接量化器23〇,接收至少一個數位瑪以產生' 一串接語音資料。 其中’另-種先前實施方式,請參考第2B圖,在一外部的第一記憶體 110已儲存了數位的第-語音資料,其中,減碼轉換器22〇對第一語音資馨 料進打減碼。量化器230連接減碼轉換器22〇,接收第一語音資料並進行 量化而產生-數位碼,該數位碼包含符號資料與數字資料。資料編碼器24〇 連接量化器230 ’接收至少一個數位碼以產生一串接語音資料。 以下列舉一範例: 接著’請參考第3圖’由類比數位轉換S 210所轉換之語音資 料’其中包含了 7筆數位的第一語音資料:11111〇11〇〇〇1〇〇〇、 1111001100001000 . 1111111100001000 . 0000000100001000 . # 0000010100001000、〇〇〇〇1〇〇〇〇〇〇〇1〇〇〇、〇〇〇〇1 i ] 1〇〇〇〇1 〇〇〇。減媽轉 換器220再將這7筆的I6bit第-語音資料轉為8 bit的有帶正、負符號的 第-語音資料。其中減碼轉換器22〇直接把16bjt的第一語音資料中的第 1 bit至第8bit的直接去掉,只保留原來的第一語音的第到第)哪的資 料’而运留下的資料’即為新的第一語音資料。所以第一語音資料最後只 留下8以的具有正、負符號的資料,且其資料範圍為-128至127。其中, 而第1bit至第7bit則代表數字資料(語音訊號的量),而第_則代表符號 4 M438765 丨㈣月K修正 是原來7筆讎的第-語音資料共有112bit,經過減瑪轉換 咖t的資料轉換成只有7筆8bit共56bit的第—語音資料。再糊量化器 230的量化結果將每筆8bit資料(表格碼)變成獅的索引碼資料,最後,7 筆5bit的資料量是35 bi卜由此可知,我們由原先的扣㈤的資料量經過 減碼轉麟220與量化㈣G,最後魏只有35bit崎料量。之後,再加 上1〇bit的訊框標記資料606,現在總資料量共為45盼。 由以上的先前技術可知’在做語音量化編碼時,其量化的每筆數位碼 _ 含有符號碼與數字瑪,而每筆數位碼都包含有符號碼,無形中會多浪費儲 存的資料量。所以為了減少浪費儲存的資料量,實有必要提出一種新的架 構來減少儲存的資料量。 【新型内容】 ▲本創作提供-種音訊量化編碼裝置,運用—記憶體以進行訊號編碼, 記憶體紀錄有複數個數位第一語音資料,包含:訊號分割器、量化器與資 料編。—訊號分,讀取該錄位第—語音資料並進行複數個零交 馨魅判斷而依序產生複數個第一符號資料,並將該些數位第一語音資料切 割為複數個訊框;一量化器,連接訊號分割器,接收每個訊框所對應的該 些數位第-語音資料與第一符號資料,並將每次所接收之訊框所所對應之 該些數位第-語音·量化後產生複數㈣—數字資料,並依據訊框量化 :=雜應產生—第—訊框標記資料;及—資料編,連接量化器與訊 L ’接收量化器對每個訊框所產生該些第一數字資料、第—符號資 料八第訊框;^記資料並編碼成一第一編碼資料串。 本創作又提供—種音訊量化解碼裝置,運用一記憶體以進行訊號解 M438765 T 庄 t 次’記憶體紀錄有—第二編爾串,包含:資料解碼器與反量Wr-貝科解碼器’連接記憶體,讀取第二編碼資料串並進行解碼而產生複數個 第-解碼資辦,每_二解碼_包含:—第二訊框標_、一第 一符娜、伽第反量峨麵職,接收 _解碼貝料串並依據第二訊框標記資料、第二符號資料之值進行該些 第二數字資料之反量化而依序產生複數個數位第二語音資料。 本創作提出-種更有效率輪_,現有的編碼裝置其每一筆量化 的數位碼都包含有符麵,無形巾會多㈣儲存的麟量。本創作只提出 只用-個符號碼,並串聯概個數字胁,在轉解析度的前提下可以減 少儲存的資料量’而在追求音質的前提下可以透過微幅增加資料量來達到 顯著提升音質的功效。 為讓本創作之上述和其他目的、特徵、和優點能更明顯易懂,下文特 舉數個較佳實_,並齡所關式,作詳細削如下: 【實施方式】 請參考第5A目’係、為本創作的音訊量化編補組2·之實施例,包 含:量化ϋ 230、訊號分廳25Q與㈣編碼器2.訊號分翻25〇包 括了-個暫存H 251,訊號分割器25Q從第—記憶體11〇讀取所儲存之數 位第-語音資料並進行-零交越點判斷而依序產生複數個第—符號資料 612 ’並將數位第一語音資料切割為複數個訊框。量化器23〇連接訊號分割 器250,依序依據訊號分割$ 25〇所切割的訊框#中所對應的多個數位第 一浯音貢料與與相對應的第一符號資料,並將訊樞所對應之多個數位第一 語音資料4化後_對-對應產生第-數字資料,並依據此次訊框所量化之 8 M438765 結果產生第一訊框標記資料606。資料编碼器240則連接量化 號分割器250 ’接收第一數字資料、第一符號資料與訊框標記資料並將其 編碼成第一編碼資料串,每個第一編瑪資料串包含第一訊框標記資料、第 一符號資料、多個第一數字資料。之後,再將第一编碼資料串儲存之第二 記憶體310中。 實務上’第一記憶體110與第二記憶體31〇可以是一個相同記憶體當 中的不同區塊。 接著,請參考第5B圖,係為本創作的音訊量化編碼模組2〇〇B之實施 例。第5B圖與第5A圖中主要的差異為’第5B圖中之量化器23〇每次所 讀取之訊框所對應的多個數位第-語音資料係直接從第一記憶體11〇讀取 後並進行量化。而第5A圖中,量化n 23Q每次所讀取之訊框所對應的多個 數位第-語音資料先由第-記憶體11()讀取後放置於訊號分割器25〇之暫 存器251 t,再由訊號分割器25〇之暫存器251讀取數位第一語音資料後 進行量化。 接著,請參考第6A ,其為本創作的音訊量化編碼模組2〇〇c之實施 例。其為於第5A圖的實施例中,增加了一個減碼轉換器22〇。減碼轉換器 220連接第-記憶體110與訊號分廳25〇之間,對暫存於第—記憶體训 當中-個訊框的所有第-語音資料進行減碼之動作,再儲存於暫存器⑸ 當中。 接著,第6B圖’其為本創作的音訊量化編碼模組2〇〇d之實施例。其 為於第5B圖的實施例中,增加了一個減碼轉換器22〇。減碼轉換器挪 連接於第-記憶體no與訊號分割器25〇、量化器23〇之間,對儲存於第 9 M438765 jms ' 一記憶體中110的第一語音資料進行減碼之動作,再傳送至量化~” 其中,量化器230包含··控制單元及向量單元,控制單元依據每個訊 框所對應之所有的第一語音資料或經減碼之第一語音資料並計算量化誤差 等以供量化表的選擇,並產生訊框標記資料。向量單元接收第一語音資料 並進行量化表的查尋而對應產生數字資料β 本創作的減碼轉換器220並不侷限只有16bit減少為8bit,亦可由16bit 變成i〇bit,又或者24bl.t減少為i2bit,本創作並沒有限定格式完全依據V. New type description: [New technical field] This creation is a kind of quantization device, especially for an audio quantization device. [Prior Art] The speech signal is originally analog signal, which will be distorted after digitization and compression. In terms of compression ratio, signal distortion is large, but the required transmission code rate is low. Therefore, in the case of insufficient transmission bandwidth, 'below the cake under the rhyme call content', the limbs with high heights are usually selected. If there is no problem with the transmission bandwidth, it is generally better to use the G711 protocol with less signal distortion. Please refer to FIG. 1 , which is a prior art speech encoding and decoding system diagram, including: a voice input signal 100, a voice encoder 200, a memory 300, a voice decoder 400, and a voice output signal 500. Among them, the voice input signal is a true sound of the segment, which is an analog signal. For example, if the speech encoder 200 is a 16-bit mono, if the sampling rate is 8 KHz per second, the data amount is 12 sec per second, and when the speech input signal is input to the speech coder 2 〇〇, After the voice input, it will be sampled in the 128kbit channel surface of the Lin seconds. After the dust is encoded, it will be stored in the memory training +. The vocoder 2 (five) is actually a compressor. In practical applications, in order to reduce the usage of memory 3 (5), the tree generally compresses the 16 ❾ ❾ voice data into lower resolution data (such as 5bit or 4bit) and exists in the memory to effectively reduce the memory. 300 usage. Finally, the speech decoder 4 解读 interprets the lower resolution data after the memory 300 has been compressed and converts it into a mono speech material' having one and converts it into a speech output signal 500. Next, the reference to Fig. 2A is a detailed block diagram of the prior art speech coder. The voice knitting machine 200 includes: an analog digital converter 21 〇, a down code converter 22 量, a quantity 3 M438765 I 月 month f day double 2 230 and data editing · 24Ge, the digital converter plus receiving class The first voice data converted into a digit by inputting a signal. The subtractive conversion $22〇 connection analog digital converter 210' de-codes the first speech data. The quantizer 23 is coupled to the down-converter, receives the first-speech data and quantizes to generate a digital code containing the symbol data and the digital data. The data encoder 240 is coupled to the quantizer 23A to receive at least one digit to generate 'a series of speech data. In the other previous embodiment, please refer to FIG. 2B, in which an external first memory 110 has stored digital first-speech data, wherein the down-converter 22〇 enters the first voice. Reduce the code. The quantizer 230 is coupled to the down-converter 22A, receives the first speech data and quantizes to produce a-digit code containing symbol data and digital data. The data encoder 24 〇 connection quantizer 230' receives at least one digital code to generate a concatenated speech material. An example is given below: Next, please refer to Figure 3 for the speech data converted by the analog-to-digital conversion S 210, which contains the first speech data of 7 digits: 11111〇11〇〇〇1〇〇〇, 1111001100001000. 1111111100001000 . 0000000100001000 . # 0000010100001000,〇〇〇〇1〇〇〇〇〇〇〇1〇〇〇,〇〇〇〇1 i ] 1〇〇〇〇1 〇〇〇. The reduction mother converter 220 converts the seven I6bit first voice data into an 8-bit first voice data with positive and negative signs. The down code converter 22 directly removes the first bit to the eighth bit of the first voice data of the 16bjt, and retains only the data of the first to the first voice of the original voice. This is the new first voice material. Therefore, the first speech data only leaves 8 positive and negative symbols, and the data range is -128 to 127. Among them, the first bit to the seventh bit represent digital data (the amount of voice signals), and the first _ represents the symbol 4 M438765 丨 (four) month K correction is the original 7-note 第 first voice data a total of 112bit, after the reduction of the conversion of coffee The data of t is converted into the first voice data of only 7 8 bits and 56 bits. The quantization result of the re-quantizer 230 converts each 8-bit data (form code) into the index data of the lion. Finally, the data amount of the 7-bit 5-bit is 35 bib. From this, we can know that the amount of data of the original buckle (five) has passed. Reduced code to Lin 220 and quantified (four) G, and finally Wei only 35bit raw material. After that, add 1 〇 of the frame mark data 606, and now the total amount of data is 45. It can be known from the above prior art that when performing speech quantization coding, each quantized digital code _ contains a symbol code and a digital code, and each digital code contains a symbol code, which inevitably wastes the stored data amount. Therefore, in order to reduce the amount of data that is wasted, it is necessary to propose a new architecture to reduce the amount of data stored. [New Content] ▲This creation provides a kind of audio quantization and coding device, which uses memory to encode signals. The memory records a plurality of digital first speech data, including: signal divider, quantizer and information. - the signal number, reading the first voice data of the record and performing a plurality of zero-crossing enchantment judgments to sequentially generate a plurality of first symbol data, and cutting the digital first voice data into a plurality of frames; The quantizer is connected to the signal divider, and receives the digital-speech data and the first symbol data corresponding to each frame, and quantizes the digital-speech-quantization corresponding to each received frame After the generation of complex (four) - digital data, and quantized according to the frame: = miscellaneous should be generated - the first frame of the mark data; and - data compiled, connected to the quantizer and the signal L 'receiver quantizer generated for each frame The first digital data, the first symbol data, the eighth frame; the data is encoded and encoded into a first encoded data string. This creation also provides a kind of audio quantization decoding device, using a memory for signal solution M438765 T Zhuang t times 'memory record has - the second series, including: data decoder and inverse Wr-Bec decoder 'Connecting the memory, reading the second encoded data string and decoding to generate a plurality of first decoding tasks, each _ two decoding _ includes: - the second frame label _, a first symbol, gamma inverse峨 职 , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , This creation proposes a more efficient round. In the existing encoding device, each quantized digit code contains a facet, and the invisible towel will have more (4) stored volumes. This creation only proposes to use only one symbol code, and a series of digital threats, which can reduce the amount of stored data under the premise of turning the resolution', and can achieve a significant increase by slightly increasing the amount of data under the premise of pursuing sound quality. The effect of sound quality. In order to make the above and other objects, features, and advantages of the present invention more obvious and easy to understand, the following are a few of the best _, and the age is closed, and the detailed cutting is as follows: [Embodiment] Please refer to the 5A 'Department, the embodiment of the audio quantization and editing group 2 of the creation, including: quantization ϋ 230, signal sub-office 25Q and (four) encoder 2. signal sub-turn 25 〇 includes a temporary storage H 251, signal segmentation The device 25Q reads the stored digital first-speech data from the first memory 11〇 and performs a zero-crossing point determination to sequentially generate a plurality of first symbol data 612′ and cut the digital first speech data into a plurality of Frame. The quantizer 23 is connected to the signal divider 250, and sequentially divides the plurality of digits of the first arpeggio and the corresponding first symbol data corresponding to the frame #cut by the signal according to the signal. The plurality of digits of the first voice data corresponding to the pivot is _--corresponding to generate the first digital data, and the first frame marker data 606 is generated according to the 8 M438765 result quantized by the frame. The data encoder 240 is coupled to the quantized number divider 250 ′ to receive the first digital data, the first symbol data and the frame marker data and encodes the first encoded data string into a first encoded data string, each first encoded data string comprising the first Frame mark data, first symbol data, and multiple first digital materials. Thereafter, the first encoded data string is stored in the second memory 310. In practice, the first memory 110 and the second memory 31 can be different blocks of the same memory. Next, please refer to FIG. 5B, which is an embodiment of the audio quantization coding module 2〇〇B of the present invention. The main difference between FIG. 5B and FIG. 5A is that the plurality of digit-the speech data corresponding to the frame read by the quantizer 23 in FIG. 5B is directly read from the first memory 11 . Take it back and quantify it. In FIG. 5A, the quantized n-bit data corresponding to each frame read by n 23Q is read by the first memory 11 () and then placed in the register of the signal divider 25 251 t, and then the digital voice data is read by the register 251 of the signal divider 25 to perform quantization. Next, please refer to Section 6A, which is an embodiment of the audio quantization coding module 2〇〇c of the present invention. In the embodiment of Fig. 5A, a down code converter 22 is added. The down code converter 220 is connected between the first memory 110 and the signal distribution room 25〇, and performs all the first voice data temporarily stored in the first memory training frame to be decoded, and then stored in the temporary memory. The memory (5) is in the middle. Next, Fig. 6B is an embodiment of the audio quantization coding module 2〇〇d of the present invention. In the embodiment of Fig. 5B, a down code converter 22 is added. The down-converter is connected between the first-memory no and the signal splitter 25A and the quantizer 23A, and performs the action of subtracting the first voice data stored in the memory of the first M438765 jms' memory. Then, the quantizer 230 includes a control unit and a vector unit. The control unit calculates the quantization error according to all the first voice data or the subtracted first voice data corresponding to each frame. For the selection of the quantization table, and generating the frame mark data, the vector unit receives the first voice data and performs the search of the quantization table to generate the digital data. The code reduction converter 220 of the present invention is not limited to only 16 bits and is reduced to 8 bits. It can also be changed from 16bit to i〇bit, or 24bl.t can be reduced to i2bit.
系統的設計來加以選定。 I 第扣曰為料可為經過減碼後的資料,例如,由減碼為丨丨或 _t。在本創作的一些實施例中,當多個第一語音資料為負值時,將數位 碼600當中代表正、負符號的符號資料省略並整合至訊框標記資料之後, 並形成-個_數字資料614,其僅包含了代表聲音資料的内容,而不包 3正負的貝訊’如第7圖所示者。如此’可減少先前技術當中的數位碼 600的位το數’例如,原先,減少為伽,以降低資料量。或者,增加 數位碼6〇0的位元數,例如,由原先數位碼當中的_當中只占了韻的 # 數字資料’在本創作中,增加·,也就是_的數字資料,以提高編碼 資料的解析度。 -月參考第7圖’其為本創作之串接語音資料之資料結構示意圖。個別 的串接θ貝料620、啦均包括了訊框標記資料6〇6、符號資料M2與 夕筆數子資料州。換句話說,每筆串接語音資料紗於訊框標記資料咖 與符號資料612,而終於下一筆訊框標記資枓6〇6之前。串接語音資料 622的長度取決於兩個零交越點的長度當中的數字資料的點數,亦 10 M438765 食P ’訊框標記資料606+符號資料612+兩個零交越點的點數χ 長度(例如’ 4bit或者5bit)。 換句話說’該符號資料抓的正號或負號,係利用訊號分割器判斷該 零交越點的產生,而零交越點的判斷,則.是依據連續二個第一語音資料的 資料變化而定。當本創作的訊號分割器25〇接收到接續—個正第—語音資 料與-個負第-語音資料時(無論何者先出現),即可判斷有零交越點,訊號 分割器250就會產生代表零交越點產生的符號資料612,以提供給資料編The system is designed to be selected. I The deduction can be a reduced data, for example, reduced to 丨丨 or _t. In some embodiments of the present invention, when the plurality of first voice data is a negative value, the symbol data representing the positive and negative symbols in the digit code 600 is omitted and integrated into the frame mark data, and a number of_ The data 614, which only contains the content representing the sound data, does not include the positive and negative of the Beixun 'as shown in Figure 7. Thus, the number of bits τ of the digit code 600 in the prior art can be reduced, for example, originally reduced to gamma to reduce the amount of data. Or, increase the number of bits of the digit code 6〇0, for example, from the _ of the original digit code, which only accounts for the ## data of the rhyme. In this creation, the digital data of _ is added to improve the encoding. The resolution of the data. - month reference to Figure 7 is a schematic diagram of the data structure of the serialized voice data of the creation. The individual tandem θ bead material 620 includes the frame mark data 6〇6, the symbol data M2 and the eve pen number sub-data state. In other words, each string of voice data is spliced to the frame tag data coffee and symbol data 612, and finally the next frame tag is 枓6〇6. The length of the concatenated speech data 622 depends on the number of points of the digital data among the lengths of the two zero crossing points, and also 10 M438765 food P 'frame mark data 606 + symbol data 612 + points of two zero crossing points长度 Length (eg '4bit or 5bit). In other words, the positive or negative sign of the symbol data is used to judge the generation of the zero crossing point by using the signal divider, and the judgment of the zero crossing point is based on the data of the two consecutive first voice data. Depending on the change. When the signal splitter 25 of the present creation receives the continuation-positive-speech data and the -negative-speech data (whenever it appears first), it can be judged that there is a zero crossing point, and the signal divider 250 will Generating symbolic data 612 representing the zero crossing point to provide a data compilation
碼器240。而符號資料612的正號,可以〇來代表,而符號資料612的負 號,可以1來代表。例如:第—個第—語音資料為A,第二個第—語音資 料為B,當A<0’且B>=〇時’訊號分割器250則產生符號資料612為”〇,,(正 號),即第-語音資料為由負轉正的情形,亦即,後續出現的第一語音資料 將都為正;當A>=〇 ’且B<0時,訊號分割器25〇則產生符號資料612 為Ί”(負號),即第-語音資料為由正轉負的情形,亦即,後續出現的第一語Coder 240. The positive sign of symbol data 612 can be represented by 〇, and the negative sign of symbol data 612 can be represented by 1. For example, the first-the first voice data is A, the second first voice data is B, and when A<0' and B>=〇, the signal splitter 250 generates the symbol data 612 as "〇,, ( No.), that is, the case where the first voice data is negatively positive, that is, the subsequent first voice data will be positive; when A>=〇' and B<0, the signal splitter 25〇 generates symbols The data 612 is Ί" (negative sign), that is, the case where the first-speech data is positively negative, that is, the first language that appears subsequently
音資料將都為I上述只是本創作實施零交越點判斷的—實施例,本創作 不倡限此種方式。 範例一: 本實施例魏明在代表語音訊號的數字資料編碼長度固定為_的情 況。訊框標記資料6〇6所對應的量化表可以有2個或2個以上,此時,訊 框標記資料606須採用至少,個bit來指示採用哪個表。例如,當只用到二 個量化表的狀況下,訊框標記資料6〇6可以設定表2(本實施例的第1個表) 所對應的表值為,,〇,,,而表3(本實施例的第2個表)的表值可設定為”1,,。當 採用5個量化表時,此時,訊框標記資料6㈤就需要3bit,分別對映的表 11 M438765 丨0Μ月十日f ,: 值為〇〇Q、0〇1、Q1Q、011、1(X)等。本創作可對映的量化表數,^ 一個表,或者,複數個表。 在有多個量化表的情形下,—般會依據資料的量化誤差大小來決定最 適合的表。接著,請回頭參考第3圖,其為多個第一語音資料序列(七、_12、 •1、3、5、8、15、8、5),其有二個零交越點的發生。假設共有8個表可 供使用,在、經過比對篩選後,表2與表3是最合適的表可分別運用於(_6、 -12、-1)序列及(3、5、8、15、8、5)序列。適合的量化表選擇,係為熟習 該項技藝者所熟知,不再贅述。 百先’經過第-個零交越點後,先遇到(·6、_13、_υ時,先在前面取負 的符號資料612為1,再把(-6、_13、取絕對值變成(6、13、^,首先表 2的訊框標記資料606先設為_,最後再經由查表2後可以得到索引碼 為(4、8、1),最後再對映表2的數字顏得到的序列為(⑴⑻、1〇〇〇、 0001)。之後,再加上訊框標記資料6〇6的值〇〇〇、符號資料612的值j。 最後得到的串接語音資料620為(〇〇〇、1、〇1〇〇、10〇〇、〇〇〇1)。 接著’經過第二個零交越點後’表3的訊框標記資料6〇6先設為Μ, 鲁 最後看(3、5、8、15、8、5) ’本符號資料612為0代表為正值,再對映表 3 ’找最接近的表格碼,此為熟習該項技藝所熟知,可以得到索引碼為〇、 2、3、5、3、2) ’最後再對映表3的數字資料的編碼為(〇〇〇1、〇〇1〇、〇〇1 ^、 0101、0011、0010)。之後,再加上訊框標記資料6〇6的值〇〇1以及符號 資料612的值0得到的串接語音資料為(001、〇、〇〇〇1、〇〇1〇、〇()11、⑴⑺、 0011、0010)» 接著’請參考第7圖’其最後再將正的與負的訊框編碼資料結合並加 12 M438765The audio data will all be I. The above is only the implementation of the zero crossing point judgment of this creation - the embodiment, this creation does not advocate this way. Example 1: In this embodiment, Wei Ming's digital data encoding length representing the voice signal is fixed to _. There may be two or more quantization tables corresponding to the frame mark data 6〇6. At this time, the frame mark data 606 shall use at least one bit to indicate which table to use. For example, when only two quantization tables are used, the frame mark data 6〇6 can set the table values corresponding to Table 2 (the first table in this embodiment) to be, 〇,,, and Table 3 The table value of (the second table in this embodiment) can be set to "1,". When five quantization tables are used, at this time, the frame mark data 6 (5) requires 3 bits, and the table 11 M438765 丨 0 On the tenth day of the month, f, the value is 〇〇Q, 0〇1, Q1Q, 011, 1(X), etc. The number of quantified tables that can be mapped by this creation, ^ a table, or a plurality of tables. In the case of a quantization table, the most suitable table will be determined according to the size of the quantization error of the data. Next, please refer back to Figure 3, which is a sequence of multiple first speech data (7, _12, • 1, 3, 5, 8, 15, 8, 5), which has two zero crossing points. It is assumed that there are 8 tables available for use. After comparison and comparison, Tables 2 and 3 are the most suitable tables. They are applied to the sequence of (_6, -12, -1) and the sequence of (3, 5, 8, 15, 8, and 5). The selection of suitable quantization tables is well known to those skilled in the art and will not be described again. First After the first zero crossing point, first encounter (·6, _13, _υ, first take the negative symbol data 612 to 1 in the front, and then (-6, _13, take the absolute value into (6, 13, ^, firstly, the frame mark data 606 of Table 2 is first set to _, and finally, after looking up the table 2, the index code is (4, 8, 1), and finally the sequence obtained by the digital face of the table 2 is ( (1) (8), 1〇〇〇, 0001). After that, the value of the frame mark data 6〇6 and the value j of the symbol data 612 are added. The finally obtained serial voice data 620 is (〇〇〇, 1). 〇1〇〇, 10〇〇, 〇〇〇1). Then 'after the second zero crossing point', the frame marking information of Table 3 is set to Μ6, and the last look of Lu (3, 5 , 8, 15, 8, 5) 'This symbol data 612 is 0 for a positive value, and then for the mapping table 3 'to find the closest form code, which is well known in the art, you can get the index code is 〇, 2, 3, 5, 3, 2) 'The code of the digital data of the final re-enactment table 3 is (〇〇〇1,〇〇1〇,〇〇1^, 0101,0011, 0010). After that, add The value of the message box 6〇6 of the upper frame is 〇〇1 and the character The concatenated speech data obtained by the value 0 of the data 612 is (001, 〇, 〇〇〇 1, 〇〇1 〇, 〇 () 11, (1) (7), 001, 0010) » Next 'Please refer to Figure 7' and finally Combine positive and negative frame-encoded data and add 12 M438765
—— · W IVW ixjx^yj - UUU1 '1111' 001 〇 0001、咖、〇〇11、〇1〇1、0011、〇_完整的語音資料序列 字串。 在則述的實施例中,透過將起始碼當中增設—個位元的符號資料,即 可賓略後續帶正負號數位碼的符號位元。當每兩個零交越點的點數越多, 可省略的資料量越多。 從另一個觀點而言,以4bit的量化編碼為例,以先前技術經過編碼後 的量化結果,每4個bit的數位碼當中’皆包含有1個符號資料,數值資料 只有3bit。而在本創作中,在暨有的4bit架構當中,則可將原先的4個bit 全部運用為數值資料。在此類查表法的應用中,可在相同的資料量的基礎 下,將查表的結果,亦即,與音訊號編碼的解析度提高將近1倍,大幅提 升語音訊號編碼的品質。—— · W IVW ixjx^yj - UUU1 '1111' 001 〇 0001, coffee, 〇〇11, 〇1〇1, 001, 〇_ complete speech data sequence string. In the embodiment described above, by adding a symbol data of one bit to the start code, the sign bit of the positive and negative digit code can be followed by the guest. When the number of points per two zero crossing points is increased, the amount of data that can be omitted is increased. From another point of view, taking 4-bit quantization coding as an example, the coded quantization result of the prior art includes one symbol data in every four bit digit code, and the numerical data has only 3 bits. In this creation, in the 4bit architecture of the cum, the original 4 bits can all be used as numerical data. In the application of such a look-up table method, the result of the look-up table, that is, the resolution of the audio signal coding can be nearly doubled on the basis of the same amount of data, and the quality of the voice signal coding is greatly improved.
表2 數字資料 索引碼 表格碼 0000 0 0 0001 1 Ί ~ 0010 2 2 0011 ~ 3 ~4 0100 4 ~6 0101 " 5 ~8 -- 01Ϊ0 " 6 9 01ΪΪ " 7 1 1Ϊ 1000 8 12 100Ϊ ----- 9 ~14~~~~' 1010 10 16~ 1011 11 17 '~~-- ΙΤόδ " 12 19~ 1101 13 ~20 Ιίΐό ~~ " 14 22 13 M438765Table 2 Digital data index code table code 0000 0 0 0001 1 Ί ~ 0010 2 2 0011 ~ 3 ~ 4 0100 4 ~ 6 0101 " 5 ~8 -- 01Ϊ0 " 6 9 01ΪΪ " 7 1 1Ϊ 1000 8 12 100Ϊ ----- 9 ~14~~~~' 1010 10 16~ 1011 11 17 '~~-- ΙΤόδ " 12 19~ 1101 13 ~20 Ιίΐό ~~ " 14 22 13 M438765
訊框切換控制碼Frame switching control code
^ ------- 訊框切換控制碼 Hi〔所揭露者,係為編 ~— 置的。卩分。接者,請參考第8圖,其說明 了本創作夕立 '月t气矛〇圃,兵說明 。立:旦,量化解碼触’可將本_所編碼的串接語音·予以解 馬曰1里化解碼模組包含:資料解喝器4扣與反量化器42〇。資料解碼 器410讀账繼31Q _二W嫌编梅生複數個 第二糾資料串,每個該第二解碼資料串包含:—第二訊框標記資料、一 第二符號資料、複數個第二數宇資料;及—反量絲咖,連接該資料解 碼器’接_些第二解碼資辨,並依據該第二訊枢標記請、該第二符 號貝料之值進行該些第二數字資料之反量化而依序產生複數個數位第二語 音資料後’儲存於第三記憶體510。 14 M438765 丨· \ 1^11 實務上’第二記憶體310與第三記憶體510可以是一個相k憶體當- 中的不同區塊。 - 例如^過寅料解碼器41Q解瑪後可得到前述範例1的語音資料序列 字串(1111、000、Ί、0100、] 〇〇〇、〇〇〇1、ii i、〇〇1、〇、〇〇〇1、〇〇1〇、 〇〇11、〇1〇1、0011、〇〇1〇)。 接著,先去掉語音資料序列字串的訊框切換控制碼1111,再接著反量 化15 420取符合資料的表2 ’且取符號資料為1,代表接下來的數字資料會 _ 疋負值,再經由表2得到索引碼為(4、8、1) ’最後對映表2則可得到表格 碼(6、12、1),由於符號資料612為]代表為負,所以得到的多個語音資 料為(·6、-12、…。相同的’第二個序列中,先先去掉語音資料序列字串的 訊框切換控制碼1川,且001代表第二個量化表(表3),符號資料M2為 〇,索引竭為(2、3、4、8、4、2),最後再個別對應表3,可個別得到表格 碼(5、8、12、25、12、5)〇最後反量器將還原資料為多個第_語音資料(_5、 12 _1、5、8、12、25、12、5)。最後’在將多個第_語音資料健存到第 • 三記憶體51〇中。 實矛力上,第-§己憶體110、第二記憶體31〇與第三記憶體⑽可以是 一個相同記憶體當中的不同區塊。 雖然本創狀雛實補娜如上舰,然織_·定本創作, 任何熟習相關技藝者,在不脫離本創作之精神和範圍内,當可作些許之更 動與潤飾’因此本創作之專利保護範圍須視本說明書所附之申請專利範圍 所界定者為準。 【圖式簡單說明】 15 I啦q月、雙乎 第1圖係為先前語音編瑪與解碼系統圖(先前技術). 第2A圖係為先前語音編碼器之功能方塊圖之第一實施例(先前技術” 第2B圖係為先前語音編碼器之功能方塊圖之第二實施例(先前技術); 第3圖係為先前類比數位取樣圖(先前技術); 第4圖係為先前串接語音資料圖(先前技術); 第5A圖係為本創作之語音編碼器之功能方塊圖之第一實施例; 第5B圖係為本創作之語音編碼器之功能方塊圖之第二實施例; 第6A圖係為本創作之語音編碼器之功能方塊圖之第三實施例; 鲁 第6B圖係為本創作之語音編碼器之功能方塊圖之第四實施例; 第7圖係為本創作之串接語音資料之實施例圖;及 第8圖係為本創作之語音解码器之功能方塊圖。 【主要元件符號說明】 100 语音輸入訊號 110 第一記憶體 200 語音編碼器 210 類比數位轉換器 220 減碼轉換器 230 量化器 240 資料編碼器 250 訊號分割器 251 暫存器 300 記憶體 M438765^ ------- Frame switching control code Hi [disclosed, is the editor. Score. Receiver, please refer to Figure 8, which illustrates the creation of the Xi Li 'month t gas spear, soldiers description. Vertical: Once the quantized decoding touches, the serialized speech encoded by this _ can be solved. The 曰1 里 解码 decoding module includes: data decanter 4 deduction and inverse quantizer 42 〇. The data decoder 410 reads the following 31Q _ two W syllabus Meisheng plural second correction data strings, each of the second decoded data strings includes: - the second frame mark data, a second symbol data, a plurality of second a number of data; and - the amount of silk coffee, connected to the data decoder 'connected to the second decoding identification, and according to the second pivot mark, the second symbol of the value of the second material to carry out the second number The inverse quantization of the data sequentially generates a plurality of digits of the second voice data and is then stored in the third memory 510. 14 M438765 丨· \ 1^11 In practice, the second memory 310 and the third memory 510 may be different blocks of a phase memory. - For example, after the decoding of the decoder 41Q, the speech data sequence string of the foregoing example 1 can be obtained (1111, 000, Ί, 0100, ] 〇〇〇, 〇〇〇 1, ii i, 〇〇 1, 〇 , 〇〇〇 1, 〇〇 1 〇, 〇〇 11, 〇 1 〇 1, 0101, 〇〇 1 〇). Next, first remove the frame switching control code 1111 of the speech data sequence string, and then inversely quantize 15 420 to take the table 2' of the matching data and take the symbol data as 1, indicating that the next digital data will be _ negative value, and then The index code obtained by Table 2 is (4, 8, 1) 'The final mapping table 2 can obtain the table code (6, 12, 1), and since the symbol data 612 is negative, the obtained speech data is obtained. For (.6, -12, .... the same 'in the second sequence, first remove the frame switching control code of the speech data sequence string, and 001 represents the second quantization table (Table 3), the symbol The data M2 is 〇, the index is exhausted as (2, 3, 4, 8, 4, 2), and finally, corresponding to Table 3, the form code (5, 8, 12, 25, 12, 5) can be obtained individually. The meter will restore the data to a plurality of _voice data (_5, 12 _1, 5, 8, 12, 25, 12, 5). Finally, 'save multiple _ voice data to the third memory 51 In the actual spear force, the first § memory 110, the second memory 31 〇 and the third memory (10) may be different blocks in the same memory. If you are a fan of the same ship, you will be able to make a certain change and refinement without any deviation from the spirit and scope of this creation. Therefore, the scope of patent protection of this creation must be seen in this book. The scope of the patent application attached to the specification shall prevail. [Simple description of the diagram] 15 I, q month, and the first picture are the previous speech coding and decoding system diagram (prior art). Figure 2A is First embodiment of the functional block diagram of the prior speech coder (Prior Art) Fig. 2B is a second embodiment of the functional block diagram of the previous speech coder (prior art); Fig. 3 is a previous analog digital sampling diagram (Prior Art); Figure 4 is a previous concatenated speech data map (previous technique); Figure 5A is a first embodiment of a functional block diagram of the speech coder of the present invention; The second embodiment of the functional block diagram of the speech encoder; the sixth embodiment is the third embodiment of the functional block diagram of the speech coder of the present invention; Lu 6B is the functional block diagram of the speech coder of the present invention. Fourth implementation Figure 7 is a diagram of an embodiment of the serial voice data of the present creation; and Figure 8 is a functional block diagram of the speech decoder of the present invention. [Description of main component symbols] 100 voice input signal 110 first memory 200 speech encoder 210 analog to digital converter 220 down code converter 230 quantizer 240 data encoder 250 signal divider 251 register 300 memory M438765
310 第二記憶體 400 語音解碼器 410 資料解碼器 420 反量化器 500 語音輸出訊號 510 第三記憶體 600 數位碼 606 訊框標記貢料 612 符號資料 614 數字資料 624 第一編碼資料串 626 第一編碼資料串 700 音訊編解碼器310 second memory 400 speech decoder 410 data decoder 420 inverse quantizer 500 speech output signal 510 third memory 600 digital code 606 frame mark tribute 612 symbol data 614 digital data 624 first encoded data string 626 first Coded data string 700 audio codec
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| US13/727,489 US9070362B2 (en) | 2011-12-30 | 2012-12-26 | Audio quantization coding and decoding device and method thereof |
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