TWI469135B - Adaptive differential pulse code modulation (adpcm) encoding and decoding method - Google Patents
Adaptive differential pulse code modulation (adpcm) encoding and decoding method Download PDFInfo
- Publication number
- TWI469135B TWI469135B TW100147910A TW100147910A TWI469135B TW I469135 B TWI469135 B TW I469135B TW 100147910 A TW100147910 A TW 100147910A TW 100147910 A TW100147910 A TW 100147910A TW I469135 B TWI469135 B TW I469135B
- Authority
- TW
- Taiwan
- Prior art keywords
- pulse code
- code modulation
- differential pulse
- encoder
- decoder
- Prior art date
Links
Description
本發明係有關於一種調適性差分脈衝碼調變編碼解碼的方法,尤其是指一種可避免因有線或無線通道不穩定所造成解碼端所解出之語音資料有劇烈變化的情況產生,以能有效減少語音資料解碼輸出之雜音與劇烈失真現象,而在其整體施行使用上更增實用功效特性之調適性差分脈衝碼調變編碼解碼的方法創新設計者。 The invention relates to a method for adapting differential pulse code modulation coding and decoding, in particular to a situation that can avoid the dramatic change of the speech data solved by the decoding end caused by the instability of the wired or wireless channel, so as to enable An innovative designer who effectively reduces the noise and severe distortion of the speech data decoding output, and adjusts the adaptive differential pulse code modulation coding and decoding method in its overall implementation.
按,在無線語音傳輸上,為在低頻寬低成本之2.4GRF上傳輸語音資料,使得須進行資料的壓縮與解壓縮;其中,一般常見之調適性差分脈衝碼調變〔Adaptive Differ ential Pulse Code Modulation;ADPCM〕編碼器(3),請參閱第五圖現有之編碼器架構圖所示,該編碼器(3)主要係包括減法器(31)、量化器(32)、反向量化器(33)、預測器(34)、步階適應器(35)與加法器(36),使得輸入訊號於與預測器(34)所預估出下一筆資料的大小經減法器(31)相減後,即產生一個誤差值,理想的情況下,該誤差值應僅為一個微量值,該誤差值再經量化器(32)以步階適應器(35),所提供的步階配合一固定的解析度條件下,予以量化後產生一壓縮輸出碼,而反向量化器(33)於收到該壓 縮輸出碼後,將會以步階適應器(35)所提供的相同步階值與相同的解析度條件,進行反向量化而送出一訊號,再透過加法器(36)與預測器(34)回送之預估值相加,以產生預測器(34)的輸入訊號,然後重覆前述與輸入訊號相減、對誤差值量化與反量化驗證步驟,直到結束為止。 According to the wireless voice transmission, the voice data is transmitted on the 2.4GRF with low frequency and low cost, so that the data compression and decompression are required; among them, the commonly used adaptive differential pulse code modulation [Adaptive Differential Pulse Code] Modulation; ADPCM] Encoder (3), please refer to the figure of the existing encoder architecture shown in Figure 5, the encoder (3) mainly includes a subtractor (31), a quantizer (32), and an inverse quantizer ( 33), the predictor (34), the step adaptor (35) and the adder (36), such that the input signal is subtracted from the magnitude of the next data estimated by the predictor (34) by the subtractor (31) After that, an error value is generated. Ideally, the error value should be only a small value, and the error value is further fixed by the quantizer (32) with the step adaptor (35). Under the resolution condition, it is quantized to generate a compressed output code, and the inverse quantizer (33) receives the pressure. After the output code is reduced, the phase synchronization value and the same resolution condition provided by the step adaptor (35) are inversely quantized to send a signal, and then passed through the adder (36) and the predictor (34). The feedback values of the loopback are added to generate the input signal of the predictor (34), and then the steps of subtracting the input signal from the input signal and quantizing and dequantizing the error value are repeated until the end.
然而,上述調適性差分脈衝碼調變編碼器,雖可達到對語音資料進行壓縮編碼之預期功效,但亦在整體施行使用上發現,經由ADPCM解壓縮存在著音幅滯留的效應,將會使得音樂的音量振幅,在下一個音樂框進行解壓時,會保持著前一個量,因此會一段時間內維持著大聲,或一段時間內維持著小聲,造成接收端解壓縮時預測值誤差太大,而發生聲音音量時大、時小或爆音、雜音等問題發生,致令其在整體編碼設計上仍存在有相當多改進空間。 However, the above-mentioned adaptive differential pulse code modulation encoder can achieve the expected effect of compressing and encoding the speech data, but it is also found in the overall implementation that the effect of amplitude retention is caused by the decompression of ADPCM. The volume amplitude of the music will remain at the previous amount when decompressed in the next music frame, so it will remain loud for a period of time, or keep a small sound for a period of time, causing the prediction value error when the receiver is decompressed too much. However, problems such as large, small or popping sounds, and murmurs occur when the volume of the sound occurs, so that there is still considerable room for improvement in the overall coding design.
緣是,發明人秉持多年該相關行業之豐富設計開發與實際製作經驗,針對現有之ADPCM編解碼方法再予以研究改良,提供一種調適性差分脈衝碼調變編碼解碼的方法,以期達到更佳實用價值性之目的者。 The reason is that the inventor has been rich in design and development and practical production experience of the relevant industry for many years, and has made research and improvement on the existing ADPCM codec method, and provides a method of adaptive differential pulse code modulation and decoding to achieve better and practical. The purpose of value.
本發明揭示之調適性差分脈衝碼調變編碼解碼的方法,其於編碼器令輸入訊號與預測器所預估出下一筆資料的大小經減法器相減後,產生一個誤差值,該誤差值再經量化器產生一調適性差分脈衝碼調變壓縮編碼訊號,主要係於量化器連接有量化步階單元,以編碼器步階索引表,配合編碼器步階表進行前置解壓縮,以輸出步階索引值〔index〕至前一訊號步階索引值之延遲暫存器,且步階索引值至前一訊號步階索引值之延遲暫存器延遲一時間,以於下一調適性差分脈衝碼調變壓縮編碼訊號輸出時,同步輸出前一訊號步階索引值;上述編碼輸出訊號,經由解碼器之反向量化器與 解碼器步階索引表,於解碼器步階索引表,產生指數輸入多工器與所輸入前一訊號步階索引值進行多工處理後,經解碼器步階表提供步階索引值於反向量化器進行反向量化後,再送出一訊號至預測器,以進行解壓縮與解碼動作;藉此,可避免因有線或無線通道不穩定所造成解碼端,所解出之語音資料有劇烈變化的情況產生,以能有效減少語音資料解碼輸出之雜音與劇烈失真現象。 The method for adapting differential pulse code modulation coding and decoding according to the present invention is characterized in that the encoder causes the input signal and the magnitude of the next data estimated by the predictor to be subtracted by the subtractor to generate an error value, and the error value is obtained. Then, the quantizer generates an adaptive differential pulse code modulation compression coding signal, which is mainly connected to the quantization step unit by the quantizer, and uses the encoder step index table to perform pre-decompression with the encoder step table. Output the step index value [index] to the delay register of the previous signal step index value, and the delay index of the step index value to the previous signal step index value is delayed by one time for the next adaptation. When the differential pulse code modulation compression coded signal output is output, the previous signal step index value is synchronously output; the above coded output signal is passed through the inverse quantizer of the decoder and a decoder step index table, in the decoder step index table, generating an exponential input multiplexer and performing multiplex processing on the input previous step index value, and providing the step index value in the decoder step table After the vectorizer performs inverse quantization, a signal is sent to the predictor for decompression and decoding operations; thereby, the decoding end caused by the instability of the wired or wireless channel can be avoided, and the decoded speech data is severe The change occurs to effectively reduce the noise and severe distortion of the speech data decoding output.
根據上述構想,本發明所述之調適性差分脈衝碼調變編碼解碼的方法,主要係由編碼器輸入原始脈衝編碼調變〔PCM〕資料經編碼處理後,輸出目前音訊框的調適性差分脈衝碼調變壓縮編碼值〔ADPCM CODE〕、前一個音訊框的步階索引值〔Previous index〕與控制資訊予以合併成單一封包,經過有線或無線的通訊方式,傳送到解碼器來進行解碼,以還原出原始脈衝編碼調變〔PCM〕資料。 According to the above concept, the method for adapting the differential pulse code modulation and decoding according to the present invention mainly comprises the following steps: the encoder inputs the original pulse code modulation (PCM) data, and then outputs the adaptive differential pulse of the current audio frame. The code modulation compression code value (ADPCM CODE), the previous index of the previous audio frame and the control information are combined into a single packet, and transmitted to the decoder for decoding by wired or wireless communication. The original pulse code modulation (PCM) data is restored.
根據上述構想,本發明所述之調適性差分脈衝碼調變編碼的方法,包含下列步驟:編碼器對輸入資料進行壓縮編碼,產生一編碼訊號;量化步階單元同時接收差分信號進行前置解壓縮,以編碼器步階索引表配合編碼器步階表,以產生一步階索引值至前一訊號步階索引值之延遲暫存器;前一訊號步階索引值之延遲暫存器延遲一單位時間,輸出步階索引值;產生本次調適性差分脈衝碼調變壓縮編碼訊號,以及前一訊號步階索引值之編碼器,傳送訊號輸出至解碼器。 According to the above concept, the method for adapting differential pulse code modulation coding according to the present invention comprises the following steps: the encoder compresses and encodes the input data to generate an encoded signal; and the quantization step unit simultaneously receives the differential signal for pre-solution. Compression, the encoder step index table is matched with the encoder step table to generate a one-step index value to the delay register of the previous signal step index value; the delay register delay of the previous signal step index value is one Unit time, output step index value; generate the adaptive differential pulse code modulation compression coded signal, and the encoder of the previous signal step index value, and transmit the signal to the decoder.
根據上述構想,本發明所述之調適性差分脈衝碼調變解碼的方法,為在進行第一根音訊PCM的還原時,必須載入前一訊框的最後一筆步階索引值進行解碼,讓步階索引值能延續前一個訊框的最後一個音訊PCM所產生的步階索引值,因而避免解碼端所解出 之語音資料有劇烈變化的情況產生,以維持解碼後之音質穏定。 According to the above concept, the method for adapting the differential pulse code modulation and decoding according to the present invention is to load the last step index value of the previous frame for decoding when the first audio PCM is restored. The order index value can continue the step index value generated by the last audio PCM of the previous frame, thereby avoiding the decoding end The voice data is changed drastically to maintain the decoded sound quality.
(1)‧‧‧編碼器 (1)‧‧‧Encoder
(11)‧‧‧減法器 (11)‧‧‧Subtractor
(12)‧‧‧量化器 (12) ‧ ‧ Quantizer
(13)‧‧‧量化步階單元 (13)‧‧‧Quantitative step unit
(131)‧‧‧編碼器步階索引表 (131)‧‧‧Encoder step index table
(132)‧‧‧編碼器步階表 (132)‧‧‧Encoder step scale
(14)‧‧‧反向量化器 (14)‧‧‧Reverse Quantizer
(15)‧‧‧預測器 (15)‧‧‧ predictor
(16)‧‧‧前一訊號步階索引值之延遲暫存器 (16) ‧‧‧ Delay register of the previous signal step index value
(2)‧‧‧解碼器 (2)‧‧‧Decoder
(21)‧‧‧反向量化器 (21)‧‧‧Reverse Quantizer
(22)‧‧‧解碼器步階索引表 (22)‧‧‧Decoder step index table
(23)‧‧‧預測器 (23) ‧‧‧ predictors
(24)‧‧‧多工器 (24) ‧‧‧Multiplexer
(25)‧‧‧解碼器步階表 (25)‧‧‧Decoder step scale
(3)‧‧‧編碼器 (3)‧‧‧Encoder
(31)‧‧‧減法器 (31)‧‧‧Subtractor
(32)‧‧‧量化器 (32) ‧ ‧ Quantizer
(33)‧‧‧反向量化器 (33)‧‧‧Reverse Quantizer
(34)‧‧‧預測器 (34) ‧‧‧ predictors
(35)‧‧‧步階適應器 (35) ‧ ‧ step adaptor
(36)‧‧‧加法器 (36)‧‧‧Adder
第一圖:本發明之整體架構圖 First Picture: Overall Architecture Diagram of the Invention
第二圖:本發明之編碼器架構圖 Second picture: encoder architecture diagram of the present invention
第三圖:本發明之編碼器輸送訊號示意圖 Third: Schematic diagram of the encoder transmission signal of the present invention
第四圖:本發明之解碼器架構圖 Fourth figure: decoder architecture diagram of the present invention
第五圖:現有之編碼器架構圖 Figure 5: Existing encoder architecture diagram
為令本發明所運用之技術內容、發明目的與達成之功效,能有更完整且清楚的揭露,茲於下詳細說明之,並請一併參閱所揭之圖式及圖號: For a more complete and clear disclosure of the technical content, the purpose of the invention and the effect achieved by the present invention, it will be explained in detail below, and please refer to the illustrated figure and figure number:
首先,請參閱第一圖本發明之整體架構圖所示,本發明主要係令編碼器〔ADPCM Encoder〕(1)分別對解碼器〔ADPCM Decoder〕(2)輸入調適性差分脈衝碼調變壓縮編碼訊號與前一訊號步階索引值〔Previous index〕;其中:該編碼器〔ADPCM Encoder〕(1),請一併參閱第二圖本發明之編碼器架構圖所示,其主要係於一減法器〔Signal Differ〕(11)連接有量化器〔Quantizer〕(12),量化器(12)分別連接有量化步階單元(13)與反向量化器〔Inverse Quantizer〕(1 4),該量化步階單元(13)於量化器(12)後端,連接有編碼器步階索引表〔Index table〕(131),該編碼器步階索引表(131)後端,分別回授至其本身、連接有編碼器步階表〔Step table〕(132)與前一訊號步階索引值之延遲暫存器〔Previous Index delay Holder(Z-1)〕(16),該編碼器步階表(132)分別連接至量化器(12)與反向量化器(14),該反向量化器(14)連接有預測器〔Predictor〕(15),該預測器(15)亦連接至減法器(11)。 First, referring to the first figure, the overall architecture diagram of the present invention is shown. The present invention mainly enables the encoder [ADPCM Encoder] (1) to input the adaptive differential pulse code modulation compression to the decoder [ADPCM Decoder] (2). The encoded signal and the previous signal step index value [Previous index]; wherein: the encoder [ADPCM Encoder] (1), please refer to the second diagram of the encoder architecture diagram of the present invention, which is mainly used in one The subtractor (11) is connected to a quantizer (12), and the quantizer (12) is connected with a quantization step unit (13) and an inverse quantizer (Inverse Quantizer) (1). 4), the quantization step unit (13) is connected to an encoder step index table (131) at the back end of the quantizer (12), and the encoder step index table (131) has a back end, respectively The delay register (Previous Index Delay Holder (Z-1)) (16), which is coupled to itself, is connected with an encoder step table (132) and a previous signal step index value (16), the code The step table (132) is respectively connected to the quantizer (12) and the inverse quantizer (14), and the inverse quantizer (14) is connected with a predictor (15), and the predictor (15) is also connected. Connect to the subtractor (11).
如此一來,當利用本發明於壓縮編碼時,其係令脈衝編碼調變〔PCM〕輸入訊號與預測器(15),預估出下一筆資料的大小經減法器(11)相減後,所產生一個誤差值,該誤差值再經量化器(12)予以量化後,產生一調適性差分脈衝碼調變壓縮編碼值〔ADPCM CODE〕,於量化步階單元(13)以編碼器步階索引表(131),配合編碼器步階表(132)接收差分信號進行前置解壓縮處理,且由編碼器步階索引表(131)輸出一索引值〔Index〕,至前一訊號步階索引值之延遲暫存器(16)來延遲一單位時間,以於下一調適性差分脈衝碼調變壓縮編碼值〔ADPCM CODE〕輸出時,同步輸出前一訊號步階索引值,而反向量化器(14)於收到該調適性差分脈衝碼調變壓縮編碼訊號後,將會以編碼器步階索引表(131),配合編碼器步階 表(132)所提供的相同步階索引值,進行反向量化而送出一訊號至預測器(15),以產生預測器(15)的輸入訊號然後,重覆前述與輸入訊號相減、對誤差值量化與反量化驗證步驟,直到結束為止。 In this way, when the present invention is used for compression coding, the pulse code modulation (PCM) input signal and the predictor (15) are estimated, and the size of the next data is estimated to be subtracted by the subtractor (11). An error value is generated, which is quantized by the quantizer (12) to generate an adaptive differential pulse code modulated compression code value (ADPCM CODE), and the encoder step is used in the quantization step unit (13). The index table (131) cooperates with the encoder step table (132) to receive the differential signal for pre-decompression processing, and outputs an index value [Index] from the encoder step index table (131) to the previous signal step. The index value delay register (16) delays one unit time to synchronously output the previous signal step index value and the inverse vector when the next adaptive differential pulse code modulation compression code value [ADPCM CODE] is output. After receiving the adaptive differential pulse code modulated compression coded signal, the chemist (14) will use the encoder step index table (131) to match the encoder step The phase synchronization index value provided in Table (132) is inverse quantized to send a signal to the predictor (15) to generate an input signal of the predictor (15), and then repeating the subtraction from the input signal, Error value quantization and inverse quantization verification steps until the end.
而經該編碼器(1)所輸出之訊號,請一併參閱第三圖本發明之編碼器輸送訊號示意圖,即能包括由量化器(12)所量化後產生的調適性差分脈衝碼調變壓縮編碼值,及由前一訊號步階索引值之延遲暫存器(16),所輸出之前一訊號步階索引值與控制資訊;其控制資訊內包含有每個音訊框的編號、編碼器識別碼、封包流量控制訊息與跳頻資訊等相關資料,而該音訊框的編號,可用供解碼端計數封包丟失的數量,讓編號由編碼端從0~n進行編號,其中,n可為常數100以上到1000以下;而跳頻資訊,則可供編碼端告知解碼端進行跳頻。 For the signal outputted by the encoder (1), please refer to the third diagram of the encoder transmission signal of the present invention, which can include the adaptive differential pulse code modulation generated by the quantizer (12). Compressing the encoded value, and the delay register (16) of the previous signal step index value, outputting the previous signal step index value and control information; the control information includes the number of each audio frame, the encoder Identification code, packet flow control message and frequency hopping information, etc., and the number of the audio frame can be used by the decoder to count the number of packets lost, so that the number is numbered from 0~n by the encoding end, where n can be a constant More than 100 to 1000 or less; and frequency hopping information, the encoding end can be used to inform the decoder to perform frequency hopping.
另,該解碼器〔ADPCM Decoder〕(2),請一併參閱第四圖本發明之解碼器架構圖所示,這於對應編碼器(1)之量化器(12)所量化後,產生的調適性差分脈衝碼調變壓縮編碼訊號,分別設有反向量化器〔Inverse Quantizer〕(21)與解碼器步階索引表〔Index table〕(22),該反向量化器(21)連接有預測器〔Predictor〕(23),該解碼器步階索引表(22)分別回授至本身與連接有多工器〔Mux〕(24),且由編碼器(1)之前一訊號 步階索引值之延遲暫存器(16),所輸出的前一訊號步階索引值,亦連接輸入至該多工器(24),該多工器(24)連接有解碼器步階表〔Step table〕(25),該解碼器步階表(25)則連接至該反向量化器(21)。 In addition, the decoder [ADPCM Decoder] (2), please refer to the fourth diagram of the decoder architecture diagram of the present invention, which is generated by the quantizer (12) of the corresponding encoder (1). The adaptive differential pulse code modulation compression coding signal is respectively provided with an inverse quantizer (Inverse Quantizer) (21) and a decoder step index table (Index table) (22), and the inverse quantizer (21) is connected Predictor (23), the decoder step index table (22) is respectively fed back to itself and connected to the multiplexer [Mux] (24), and a signal preceded by the encoder (1) The step index value of the delay register (16), the output of the previous signal step index value is also connected to the multiplexer (24), and the multiplexer (24) is connected with the decoder step table [Step table] (25), the decoder step table (25) is connected to the inverse quantizer (21).
使得由編碼器(1)傳輸至解碼器(2)之調適性差分脈衝碼調變壓縮編碼訊號及前一訊號步階索引值,該調適性差分脈衝碼調變壓縮編碼訊號,分別輸入該解碼器(2)之反向量化器(21)與解碼器步階索引表(22),於解碼器步階索引表(22),產生指數輸入多工器(24)與所輸入前一訊號步階索引值進行多工處理後,在進行第一根音訊PCM的還原時,必須載入前一音訊框的最後一筆步階索引值進行解碼,讓索引值能延續前一個音訊框的最後一個音訊PCM所產生的步階索引值,經解碼器步階表(25)提供步階索引值於反向量化器(21)進行反向量化,而送出一訊號至預測器(23)進行解壓縮與解碼動作,以維持音量大或小,而不會產生忽大忽小之現象,因而避免解碼端所解出之語音資料有劇烈變化的情況產生,使能有效減少語音資料解碼輸出之雜音與劇烈失真現象,以維持解碼後之音質穏定。 The adaptive differential pulse code modulated compression encoded signal and the previous signal step index value transmitted by the encoder (1) to the decoder (2), the adaptive differential pulse code modulated compression encoded signal, respectively inputting the decoding The inverse quantizer (21) of the (2) and the decoder step index table (22), in the decoder step index table (22), generate the exponential input multiplexer (24) and the input previous signal step After the multiplex processing of the order index value, when the first audio PCM is restored, the last step index value of the previous audio frame must be loaded for decoding, so that the index value can continue the last audio of the previous audio frame. The step index value generated by the PCM is inversely quantized by the inverse quantizer (21) by providing the step index value by the decoder step table (25), and a signal is sent to the predictor (23) for decompression and Decoding action to maintain the volume is large or small, without the phenomenon of sudden increase and small, thus avoiding the situation that the speech data solved by the decoding end is drastically changed, so as to effectively reduce the noise and severeness of the speech data decoding output. Distortion to maintain the quality of the decoded sound stable.
藉由以上所述,本發明結構之組成與方法實施說明可知,本發明與現有結構相較之下,本發明主要係可避免因有線或無線通道不穩定,所造成解碼端所解出之語音資料有劇烈變化的情況產生,以能有效減少語音資料解碼輸出之雜音與劇烈失真現象,可多方應 用在無線音樂喇叭、數位教室無線電腦操控裝置、數位家庭無線電腦操控裝置與PUB DJ與遠端操控。 As described above, the composition and method implementation of the structure of the present invention show that, in comparison with the prior art, the present invention mainly avoids the speech that is decoded by the decoding end due to the instability of the wired or wireless channel. The data has been changed drastically, so as to effectively reduce the noise and severe distortion of the speech data decoding output, which can be multi-party Used in wireless music speakers, digital classroom wireless computer control devices, digital home wireless computer control devices and PUB DJ and remote control.
然而前述之實施例或圖式,並非限定本發明之產品結構或使用方式,任何所屬技術領域中具有通常知識者之適當變化或修飾,皆應視為不脫離本發明之專利範疇。 However, the above-described embodiments or drawings are not intended to limit the structure or the use of the product of the present invention, and any suitable changes or modifications may be made without departing from the scope of the invention.
綜上所述,本發明實施例確能達到所預期之使用功效,又其所揭露之具體方法,不僅未曾見諸於同類產品中,亦未曾公開於申請前,誠已完全符合專利法之規定與要求,爰依法提出發明專利之申請,懇請惠予審查,並賜准專利,則實感德便。 In summary, the embodiment of the present invention can achieve the expected use efficiency, and the specific method disclosed therein has not been seen in the same product, nor has it been disclosed before the application, and has completely complied with the provisions of the patent law. And the request, the application for the invention of a patent in accordance with the law, please forgive the review, and grant the patent, it is really sensible.
(1)‧‧‧編碼器 (1)‧‧‧Encoder
(2)‧‧‧解碼器 (2)‧‧‧Decoder
Claims (8)
Priority Applications (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
TW100147910A TWI469135B (en) | 2011-12-22 | 2011-12-22 | Adaptive differential pulse code modulation (adpcm) encoding and decoding method |
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
TW100147910A TWI469135B (en) | 2011-12-22 | 2011-12-22 | Adaptive differential pulse code modulation (adpcm) encoding and decoding method |
Publications (2)
Publication Number | Publication Date |
---|---|
TW201327547A TW201327547A (en) | 2013-07-01 |
TWI469135B true TWI469135B (en) | 2015-01-11 |
Family
ID=49225151
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
TW100147910A TWI469135B (en) | 2011-12-22 | 2011-12-22 | Adaptive differential pulse code modulation (adpcm) encoding and decoding method |
Country Status (1)
Country | Link |
---|---|
TW (1) | TWI469135B (en) |
Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
WO2001003317A1 (en) * | 1999-07-02 | 2001-01-11 | Tellabs Operations, Inc. | Coded domain adaptive level control of compressed speech |
US20040066319A1 (en) * | 2002-10-04 | 2004-04-08 | Hsien-Ming Chang | Method and apparatus for providing fast data recovery with adaptive pulse code modulation coding |
-
2011
- 2011-12-22 TW TW100147910A patent/TWI469135B/en not_active IP Right Cessation
Patent Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
WO2001003317A1 (en) * | 1999-07-02 | 2001-01-11 | Tellabs Operations, Inc. | Coded domain adaptive level control of compressed speech |
US20040066319A1 (en) * | 2002-10-04 | 2004-04-08 | Hsien-Ming Chang | Method and apparatus for providing fast data recovery with adaptive pulse code modulation coding |
Non-Patent Citations (1)
Title |
---|
Digital cellular telecommunications system (Phase 2+); Link Adaptation (GSM 05.09 version 8.1.0 Release 1999), European Telecommunications Standards Institute (ETSI) TS 101 709 V8.1.0 (2000-08), 650 Route des Lucioles F-06921 Sophia Antipolis Cedex - FRANCE (http://www.etsi.org) * |
Also Published As
Publication number | Publication date |
---|---|
TW201327547A (en) | 2013-07-01 |
Similar Documents
Publication | Publication Date | Title |
---|---|---|
CN101051465B (en) | Method and apparatus for decoding layer encoded data | |
TW201212006A (en) | Full-band scalable audio codec | |
EP3513406B1 (en) | Audio signal processing | |
CN108718361B (en) | Audio file playing method and wireless answering device | |
WO2011124161A1 (en) | Audio signal coding and decoding method, device, and codec system | |
TW200917764A (en) | System and method for providing AMR-WB DTX synchronization | |
JP6652123B2 (en) | Compression encoding device, compression encoding method, decoding device, decoding method, and program | |
WO2011137841A1 (en) | Method and device for compression encoding, method and device for decompression decoding, and communication system | |
US20100324911A1 (en) | Cvsd decoder state update after packet loss | |
TW200818124A (en) | Encoding an audio signal | |
JP2002221994A (en) | Method and apparatus for assembling packet of code string of voice signal, method and apparatus for disassembling packet, program for executing these methods, and recording medium for recording program thereon | |
TWI469135B (en) | Adaptive differential pulse code modulation (adpcm) encoding and decoding method | |
CN110770822B (en) | Audio signal encoding and decoding | |
WO2012048662A1 (en) | Method, device and system for data compression and decompression | |
US10375131B2 (en) | Selectively transforming audio streams based on audio energy estimate | |
US11545161B2 (en) | Wireless communication device, and method and apparatus for processing voice data | |
US20220038818A1 (en) | Optimized Audio Forwarding | |
JP4218456B2 (en) | Call device, call method, and call system | |
JP3193515B2 (en) | Voice coded communication system and apparatus therefor | |
US10742231B2 (en) | Compression/encoding apparatus and method, decoding apparatus and method, and program | |
JP4693185B2 (en) | Encoding device, program, and recording medium | |
US7929520B2 (en) | Method, system and apparatus for providing signal based packet loss concealment for memoryless codecs | |
WO2011012029A1 (en) | Multiple description audio coding and decoding method, device and system | |
WO2022037444A1 (en) | Encoding and decoding methods and apparatuses, medium, and electronic device | |
WO2023049628A1 (en) | Efficient packet-loss protected data encoding and/or decoding |
Legal Events
Date | Code | Title | Description |
---|---|---|---|
MM4A | Annulment or lapse of patent due to non-payment of fees |