WO2009089700A1 - Procédé et appareil de mise à jour de l'état d'un filtre de synthèse - Google Patents

Procédé et appareil de mise à jour de l'état d'un filtre de synthèse Download PDF

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Publication number
WO2009089700A1
WO2009089700A1 PCT/CN2008/072477 CN2008072477W WO2009089700A1 WO 2009089700 A1 WO2009089700 A1 WO 2009089700A1 CN 2008072477 W CN2008072477 W CN 2008072477W WO 2009089700 A1 WO2009089700 A1 WO 2009089700A1
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Prior art keywords
synthesis filter
state
coding rate
information
updating
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PCT/CN2008/072477
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English (en)
French (fr)
Inventor
Jinliang Dai
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Huawei Technologies Co., Ltd.
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Application filed by Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Priority to EP08860832.8A priority Critical patent/EP2101317B2/en
Priority to US12/502,589 priority patent/US8046216B2/en
Publication of WO2009089700A1 publication Critical patent/WO2009089700A1/zh
Priority to US12/815,028 priority patent/US8078459B2/en
Priority to US12/883,970 priority patent/US7921009B2/en

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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/18—Vocoders using multiple modes
    • G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients

Definitions

  • the present invention relates to the field of codec technology, and in particular, to a method and apparatus for updating a state of a synthesis filter. Background technique
  • CELP Code Excited Linear Prediction
  • the codebook is used as an excitation source, which has the advantages of low rate, high synthesized speech quality and strong anti-noise.
  • the coding technology as the mainstream at the coding rate of 4.8 to 16 kb/s is widely used.
  • Figure 1 is a block diagram of the CELP speech coding end system
  • Figure 2 is a block diagram of the CELP speech decoding technology system. As shown in FIG.
  • LPC linear prediction coding
  • G.729.1 is the latest release of a new generation of speech codec standards with layered coding that provides narrowband to wideband audio quality at bit rates ranging from 8kb/s to 32kb/s, allowing In the transmission process, the outer code stream is discarded according to the channel condition, and has good channel adaptability.
  • Figure 3 is a block diagram of the G.729.1 encoder system
  • Figure 4 is a block diagram of the G.729.1 decoder system.
  • the core layer codec of G.729.1 is based on the CELP model. It can be seen from Fig.
  • the TDAC encoder will be started to work, and the residual signal between the low sub-band input signal and the local composite signal of the CELP encoder at a code rate of 12 Kb/s, respectively.
  • the high sub-band signal for TDAC encoding As can be seen from Figure 4, the decoding end is at the decoding rate higher than 14kb/s.
  • the signal components of the high and low sub-bands are decoded separately, and then the residual signal component of the low sub-band is decoded by the TDAC decoder, and the residual signal component is added to the low-band signal component reconstructed by the CELP decoder. The final reconstructed low-band signal component.
  • the reconstructed signal component of the encoder CELP encoder is used in the TDAC encoding algorithm
  • the reconstructed signal component of the decoder CELP decoder is used in the TDAC decoding algorithm. Therefore, the synchronization between the CELP coded reconstructed signal and the decoded reconstructed signal is a prerequisite for ensuring the correctness of the TDAC codec algorithm.
  • the state of the CELP encoder and the CELP decoder must be synchronized. .
  • FIG. 5 is a schematic structural diagram of a CELP encoder in the existing G.729.
  • FIG. 6 is a schematic structural diagram of a CELP decoder in the existing G.729.1.
  • the CELP model used in the narrowband portion of G.729.1 supports 8 kb/s.
  • the 12kb/s rate the synthesis filter used to reconstruct the narrowband signal components at the encoding end retains two states, one at the 8kb/s rate and the other at the 12kb/s rate.
  • the core layer excitation signal calculated by the core layer G.729 encoder is used to excite the 8 kb/s synthesis filter, and the synthesis filter state is updated;
  • the excitation signal of the enhancement layer is used to excite the 12 kb/s synthesis filter and the synthesis filter state is updated.
  • the decoding end uses only one synthesis filter, calculates the corresponding excitation according to the received actual code stream, performs synthesis filtering, and updates the filter state.
  • the synthesis filter at both encoding rates of the encoding end uses the same filter coefficients as the decoding synthesis filter, that is, the quantized LPC coefficients.
  • the encoder uses two independent excitation synthesis modules to generate corresponding excitations, respectively synthesizes the corresponding synthesis filters, and updates the synthesis filter.
  • the decoding end uses only one synthesis filter, calculates the excitation signal based on the received parameters, performs synthesis filtering, and updates the synthesis filter. If the coding rate is not switched between 8kb/s and 12kb/s, the reconstructed signal at the codec will be completely synchronized; however, if the switching between the two rates occurs, the reconstructed signal at the codec will not guarantee that the synchronization will affect. The correctness of the codec algorithm will ultimately affect the quality of the reconstructed signal at the decoder. Summary of the invention
  • Embodiments of the present invention provide a method and an apparatus for updating a state of a synthesis filter, which are used to solve the problem that the reconstructed signal of the codec is not synchronized when converting between different coding rates in the CELP encoder in the prior art, and affects the reconstructed signal of the decoding end.
  • the quality defect realize the state synchronization of the CELP codec, and ensure the consistency of the code reconstruction end of the codec when the coding rate is switched.
  • An embodiment of the present invention provides a method for updating a state of a synthesis filter, including: exciting a synthesis filter corresponding to the first coding rate by using an excitation signal of a first coding rate, and outputting the reconstructed signal information, and updating the synthesis filter The state information of the synthesis filter corresponding to the second coding rate.
  • An embodiment of the present invention provides a synthesis filter state updating apparatus, including a plurality of synthesis filters, and a state update module, configured to synthesize a filter corresponding to the first coding rate by using an excitation signal of a first coding rate. After the excitation is performed and the reconstruction signal information is output, the synthesis filter and the state information of the synthesis filter corresponding to the second coding rate are updated.
  • the method and device for updating the state of the synthesis filter provided by the embodiment of the present invention allow an independent synthesis filter to be used at each coding rate during coding, and after each frame is encoded, not only the state of the synthesis filter corresponding to the current rate The update is performed, and the state of the synthesis filter at other rates is updated in the same manner, thereby realizing the synchronization of the state of the synthesis filter at different rates of the encoder end, and ensuring the consistency of the signal reconstruction at the codec when the coding rate is switched. Improve the quality of the reconstructed signal at the decoder.
  • Figure 1 is a block diagram of the CELP speech coding end system
  • FIG. 2 is a block diagram of the CELP speech decoding technology system
  • FIG. 3 is a block diagram of the G.729.1 encoder system
  • Figure 4 is a block diagram of the G.729.1 decoder system
  • Embodiment 7 is a flowchart of Embodiment 1 of a method for updating a state of a synthesis filter according to the present invention
  • Embodiment 8 is a flowchart of Embodiment 2 of a method for updating a state of a synthesis filter according to the present invention
  • FIG. 9 is a schematic structural diagram of a synthesizing filter state updating apparatus according to the present invention. detailed description
  • the CELP encoder used in the narrowband section supports two coding rates, including 8 kb/s and 12 kb/s; two independent synthesis filters are used for narrowband signals corresponding to the two coding rates.
  • the reconstruction of the components, and the updating of the states of the two synthesis filters are no longer performed independently, but the synthesis filter corresponding to the current coding rate is excited and outputted according to the current coding rate, using the excitation signal of the current coding rate.
  • the signal information is reconstructed, not only the state information of the synthesis filter corresponding to the current coding rate is updated, but also the state information of the synthesis filter corresponding to the other coding rate is updated.
  • the output information of the synthesis filter using 8 kb/s is updated after the status information of itself is updated.
  • the state information of the synthesis filter corresponding to the rate of 12 kb/s is updated; if the current coding rate is 12 kb/s or higher, the state information of itself is performed by using the output result information of the synthesis filter of 12 kb/s.
  • the synthesis filter status information corresponding to 8 kb/s is also updated. This ensures that even if the coding rate is switched between 8 kb/s and 12 kb/s, the state of the encoder synthesis filter is kept synchronized, and the code segment narrowband reconstruction signal component can be kept consistent.
  • FIG. 7 is a flowchart of Embodiment 1 of a method for updating a state of a synthesis filter according to the present invention.
  • the code stream that is, layer 1 in Table 1, can be encoded as follows: Step 100: performing LPC analysis on the received speech signal, obtaining spectral parameter information and coefficient information of the synthesis filter corresponding to the spectral parameter, and performing quantization and inverse quantization on the spectral parameter or the synthesis filter coefficient;
  • Step 101 Perform a synthetic analysis search to obtain a codebook parameter at an encoding rate of 8 kb/s, where the codebook parameter includes an adaptive codebook parameter and a fixed codebook parameter, and performs quantization and inverse quantization;
  • Step 102 according to inverse quantization
  • the obtained adaptive codebook parameter and the fixed codebook parameter synthesize an excitation signal at a rate of 8 kb/s;
  • Step 103 Excite the synthesized signal of the 8 kb/s rate inverse quantization by using the calculated core layer excitation signal, output a reconstructed signal of the narrowband signal component, and use the reconstructed signal information to update the state of the synthesis filter corresponding to the 8 kb / s rate.
  • Step 104 Update the state information of the synthesis filter corresponding to 12 kb/s by applying the state information of the synthesis filter corresponding to the updated 8 kb/s rate.
  • the state of the synthesis filter corresponding to the updated 8 kb/s rate is covered by the synthesis filter state corresponding to 12 kb/s, or the reconstruction signal synthesized by the synthesis filter corresponding to the 8 kb/s rate is directly updated in step 104 by 12 kb.
  • the voice signal received in step 100 is preprocessed.
  • the residual information is obtained according to the reconstructed signal and the preprocessed voice signal, and the residual information is subjected to perception.
  • the residual information is returned to step 101 for a synthetic analysis search. Therefore, the above synthetic analysis search process is a closed loop process.
  • Table 1 shows the frame structure of the full rate coded 20 ms frame length allocation table used.
  • Layer 1 core layer (narrowband embedded CELP, 8kb/s)
  • Narrowband enhancement layer (narrowband embedded CELP, 12kb/s) Second level fixed codebook index 13 13 13 13 52 Second stage fixed codebook symbol 4 4 4 16 Second stage fixed codebook gain 3 2 3 2 10 Correction bit (classification information) 1 1 2
  • TDBWE Layer 3 Broadband Enhancement Layer
  • Time domain envelope mean 5 5 Time domain envelope split vector 7+7 14 Frequency domain envelope split vector 5+5+4 14 Error correction bit (phase information) 7 7
  • Embodiment 8 is a flowchart of Embodiment 2 of a method for updating a state of a synthesis filter according to the present invention.
  • this embodiment introduces the coding rate to 32 kb/s.
  • the encoding process includes the following steps:
  • Step 200 performing LPC analysis on the received speech signal, obtaining spectral parameter information and coefficient information of the synthesis filter corresponding to the spectral parameter, and performing quantization and inverse quantization on the spectral parameter or the synthesis filter coefficient;
  • Step 201 Perform a synthetic analysis search to obtain a codebook parameter of the core layer, including an adaptive codebook parameter and a fixed codebook parameter, and perform quantization and inverse quantization.
  • Step 202 adaptive codebook parameters and fixed according to inverse quantization The codebook parameter synthesizes an excitation signal at a rate of 8 kb/s;
  • Step 203 Excitation of a synthesis filter of 8 kb/s by using the calculated core layer excitation signal, and updating state information of the synthesis filter;
  • Step 204 Calculate a fixed codebook parameter of the narrowband enhancement layer, perform quantization and inverse quantization, and synthesize the enhanced excitation signal by using the inverse quantized fixed codebook parameter.
  • Step 205 Excuse the enhanced excitation signal to excite the 12 kb/s synthesis filter.
  • Step 206 updating the synthesized filter state of 8 kb/s using the updated 12 kb/s synthesis filter state; and updating the updated 12 kb/s
  • the state of the synthesis filter corresponding to the rate may cover the synthesis filter state corresponding to 8 kb/s, or in step 206, the synthesis filter corresponding to the synthesis filter corresponding to the 12 kb/s rate may be directly used to update the synthesis filter corresponding to 8 kb/s. status.
  • Step 207 applying a TDBWE encoder to encode a code stream of 14 kb/s;
  • Step 208 Perform TDAC encoding together with the highband signal component by using the difference signal between the signal received in step 200 and the reconstructed signal calculated in step 205.
  • step 206 Since the decoding end uses only one synthesis filter and performs continuous updating, after the operation of step 206 is performed on the encoding end, the consistency of the narrowband signal component reconstructed in step 205 and the narrowband signal component reconstructed by the decoding end is guaranteed, and finally It will ensure the correctness of the signal reconstructed by the decoder.
  • an independent synthesis filter is used at each coding rate during encoding, and after each frame is encoded, not only the state information of the synthesis filter corresponding to the current coding rate is updated, but also The state information of the synthesis filter corresponding to the other coding rate is synchronously updated, so that the state of the synthesis filter corresponding to different coding rates at the coding end is always kept synchronized, which ensures the consistency of the reconstructed signal of the codec even when the coding rate is switched. , improve the quality of the reconstructed signal at the decoder.
  • the third embodiment of the present invention uses a DTX/CNG technique, and the frame structure of the full-rate speech frame used is as shown in Table 1.
  • the frame structure of the full-rate noise frame used is as shown in Table 3.
  • the processing method may update the state information of the synthesis filter corresponding to the encoding rate of 12 kb/s and 8 kb/s, respectively. If there is a case of switching between a noise frame and a speech frame, and encoding a speech frame with a coding rate higher than 12 kb/s, only 8 kb/s synthesis filter is used when encoding the noise frame information.
  • Synthesizing filtering in order to avoid the situation that the codec reconstructs the narrowband signal component out of synchronization, when the encoder reconstructs the noise signal, in addition to updating the state information of the used 8kb/s synthesis filter,
  • the status information of the synthesized filter of 8 kb/s is updated by using the updated state information of the synthesis filter of 8 kb/s to ensure the synchronization of the state of the synthesis filter at the encoding end, thereby ensuring the synchronization of the narrowband signal components reconstructed by the codec. Sex.
  • LSF parameter quantizer index 1 first stage LSF quantization vector 5
  • Wideband component frequency domain envelope vector 2 6 wideband component frequency domain envelope vector 3 6
  • the CELP encoder involved in the above embodiment only describes the case where the encoding rates of 8 kb/s and 12 kb/s are supported, the method for updating the state of the synthesis filter provided by the embodiment of the present invention is not limited to two encoding rates. The switching between them can also be applied to the case of more CELP coding rates, as long as the state information of the synthesis filter at each coding rate is synchronized.
  • the synthesis filter state updating device includes a plurality of synthesis filters, and further includes a state update module, configured to perform excitation on the synthesis filter corresponding to the first coding rate by using an excitation signal of a first coding rate, and output reconstruction signal information Thereafter, the synthesis filter and the state information of the synthesis filter corresponding to the second coding rate are updated.
  • the state update module may have different composition manners, for example, may include a first update submodule, configured to update, by using the reconstructed signal information, a synthesis filter corresponding to the first coding rate. And the second update submodule, configured to update state information of the synthesis filter corresponding to the second coding rate by using state information of the synthesis filter corresponding to the updated first coding rate.
  • the state update module may also be configured by the following, including a first update submodule, configured to update state information of the synthesis filter corresponding to the first coding rate by using the reconstructed signal information; and a third update submodule, And using the reconstructed signal information, updating state information of the synthesis filter corresponding to the second encoding rate.
  • FIG. 9 is a schematic structural diagram of a device for updating a state of a synthesis filter according to the present invention, specifically a schematic diagram of a structure of a CELP encoder in G.729.1, as shown in FIG. 9, for a synthesis filter having a coding rate of 8 kb/s and 12 kb/s, or Using two independent first synthesis filters 1 and second synthesis filters 2, and applying two independent first excitation signal synthesis modules 3 and second excitation signal synthesis module 4 respectively to stimulate the corresponding synthesis filter.
  • a synthetic filter of the same structure as the CELP decoder in the existing G.729.1 can be used at the decoding end.
  • the synthesis filter state updating device provided by the embodiment is used to update the state of the synthesis filter corresponding to different coding rates in the encoder by applying the state update module, thereby ensuring the state of the synthesis filter at different rates of the coding end. Synchronization ensures that the codec reconstructs the signal consistency when the coding rate is switched, and improves the quality of the reconstructed signal at the decoding end.

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Description

合成滤波器状态更新方法及装置
技术领域
本发明涉及编解码技术领域， 尤其涉及一种合成滤波器状态更新方法及 装置。 背景技术
码激励线性预测 （ Code Excited Linear Prediction; 以下简称： CELP )编 码技术是一种中低速率语音压缩编码技术， 以码本作为激励源， 具有速率低、 合成语音质量高、 抗噪性强等优点， 在 4.8〜16kb/s编码速率上作为主流的编 码技术得到广泛的应用。 图 1为 CELP语音编码端系统框图， 图 2为 CELP 语音解码技术系统框图。 如图 1所示， 输入的语音信号经过预处理之后， 进 行线性预测编码（Linear Prediction Coding; 以下简称： LPC )分析， 获得谱 参数， 谱参数对应于合成滤波器的系数； 固定码本贡献和自适应码本贡献进 行混合并作为合成滤波器的激励， 合成滤波器输出重建信号， 该信号应与图 2 中解码端的合成滤波器的输出一致； 对重建信号与预处理后的信号的残差 进行知觉加权并进行合成分析搜索， 分别搜索出自适应码本参数和固定码本 参数用于滤波器的激励。
G.729.1是最新发布的新一代语音编解码标准，该嵌入式语音编解码标准 具有分层编码的特性， 能够提供码率范围在 8kb/s〜32kb/s的窄带到宽带的音 频质量， 允许在传输过程中， 根据信道状况丟弃外层码流， 具有良好的信道 自适应性。图 3为 G.729.1编码器系统框图，图 4为 G.729.1解码器系统框图， 如图 3、 4所示， G.729.1的核心层编解码是基于 CELP模型的。 由图 3可知， 在编码速率高于 14kb/s时， 将启动 TDAC编码器开始工作， 分别对低子带输 入信号与 12Kb/s码率下 CELP编码器的本地合成信号之间的残差信号和高子 带信号进行 TDAC编码。 由图 4可知， 解码端在解码速率高于 14kb/s时， 首 先要分别解码出高、 低两个子带的信号分量， 然后由 TDAC解码器解码出低 子带的残差信号分量， 残差信号分量与 CELP解码器重建出的低带信号分量 相加即为最终重建的低带信号分量。 由于 TDAC 编码算法中用到了编码端 CELP编码器的重建信号分量， 同时 TDAC解码算法中用到了解码端 CELP 解码器的重建信号分量。 因此， CELP编码端重建信号与解码端重建信号的同 步是保证 TDAC编解码算法正确性的前提条件， 而要保证编解码端重建信号 的同步， 就要保证 CELP编码器和 CELP解码器状态的同步。
图 5为现有 G.729.1中 CELP编码器结构示意图， 图 6为现有 G.729.1中 CELP解码器结构示意图， 如图 5所示， G.729.1中窄带部分使用的 CELP模 型支持 8kb/s和 12kb/s两种速率， 编码端用于重建窄带信号分量的合成滤波 器分别保留了两种状态， 一种是 8kb/s速率下的状态， 另一种是 12kb/s速率 下的状态。 在编码端， 若当前编码速率为 8kb/s, 则使用核心层 G.729编码器 计算出的核心层激励信号对 8kb/s的合成滤波器进行激励，并更新合成滤波器 状态； 若当前编码速率等于或者高于 12kb/s, 则使用增强层的激励信号对 12kb/s的合成滤波器进行激励， 并更新合成滤波器状态。 如图 6所示， 解码 端则仅釆用一个合成滤波器， 根据接收到的实际码流， 计算出相应的激励， 进行合成滤波， 并更新滤波器状态。 编码端两种编码速率下的合成滤波器使 用与解码端合成滤波器相同的滤波器系数， 即量化后的 LPC系数。
对于 8kb/s和 12kb/s两种编码速率， 编码端应用两个独立的激励合成模 块分别生成相应的激励， 分别对相应的合成滤波器进行合成滤波， 并对合成 滤波器进行更新。 解码端仅釆用一个合成滤波器， 根据接收到的参数计算激 励信号， 并进行合成滤波， 并更新合成滤波器。 如果编码速率没有在 8kb/s 与 12kb/s之间进行切换， 则编解码端的重建信号将完全同步； 但是若发生两 种速率之间的切换， 则编解码端的重建信号将无法保证同步将影响编解码算 法的正确性， 最终将影响解码端重建信号的质量。 发明内容
本发明实施例提供一种合成滤波器状态更新方法及装置， 用以解决现 有技术中 CELP编码器中在不同编码速率之间转换时， 编解码端重建信号 的不同步， 影响解码端重建信号质量的缺陷， 实现 CELP编解码器状态同 步， 在编码速率发生切换时保证编解码端重建信号的一致性。
本发明实施例提供一种合成滤波器状态更新方法， 包括在利用第一编 码速率的激励信号对所述第一编码速率对应的合成滤波器进行激励， 输出 重建信号信息后， 更新所述合成滤波器和第二编码速率对应的合成滤波器 的状态信息。
本发明实施例提供一种合成滤波器状态更新装置， 包括数个合成滤波 器， 还包括状态更新模块， 用于在利用第一编码速率的激励信号对所述第 一编码速率对应的合成滤波器进行激励， 输出重建信号信息后， 更新所述 合成滤波器和第二编码速率对应的合成滤波器的状态信息。
本发明实施例提供的合成滤波器状态更新方法及装置， 在编码时允许 每个编码速率下使用独立的合成滤波器， 并在每帧编码完成后， 不仅对当 前速率对应的合成滤波器的状态进行更新， 还要对其它速率下的合成滤波 器的状态进行相同的更新， 进而实现编码端不同速率下合成滤波器状态的 同步， 在编码速率发生切换时保证编解码端重建信号的一致性， 提高解码 端重建信号的质量。 附图说明
图 1为 CELP语音编码端系统框图；
图 2为 CELP语音解码技术系统框图；
图 3为 G.729.1编码器系统框图；
图 4为 G.729.1解码器系统框图；
图 5为现有 G.729.1中 CELP编码器结构示意图； 图 6为现有 G.729.1中 CELP解码器结构示意图；
图 7为本发明合成滤波器状态更新方法实施例一流程图；
图 8为本发明合成滤波器状态更新方法实施例二流程图；
图 9为本发明合成滤波器状态更新装置结构示意图。 具体实施方式
下面结合附图和具体实施例进一步说明本发明实施例的技术方案。 合成滤波器状态更新方法实施例
在语音编码解码标准 G.729.1 中， 窄带部分使用的 CELP编码器支持 两种编码速率， 包括 8kb/s和 12kb/s; 对应于该两种编码速率使用两个独 立的合成滤波器进行窄带信号分量的重建， 而且对两个合成滤波器的状态 的更新不再独立进行， 而是根据当前编码速率， 在利用当前编码速率的激 励信号对所述当前编码速率对应的合成滤波器进行激励， 输出重建信号信 息后， 不但对当前编码速率所对应的合成滤波器的状态信息进行更新， 而 且还要对其它编码速率对应的合成滤波器的状态信息进行更新。 就 G.729.1中窄带部分使用的 CELP模型而言， 若当前编码速率为 8kb/s, 则 在使用 8kb/s的合成滤波器的输出结果信息对自身的状态信息进行更新之 后， 还要对编码速率为 12kb/s对应的合成滤波器的状态信息进行更新； 若 当前编码速率为 12kb/s或者更高时， 则在使用 12kb/s的合成滤波器的输 出结果信息对对自身的状态信息进行更新之后， 还要对 8kb/s对应的合成 滤波器状态信息进行更新。 这样保证即使编码速率在 8kb/s与 12kb/s之间 发生切换， 编码端合成滤波器的状态保持同步， 编解码端窄带重建信号分 量能够保持一致。
图 7为本发明合成滤波器状态更新方法实施例一流程图，如图 7所示， 若当前编码速率为 8kb/s, 则仅需要使用 G.729编码器对窄带信号分量编 码出 8kb/s的码流， 即表 1中的层 1即可， 编码流程如下： 步骤 100, 对接收到的语音信号进行 LPC分析， 获得谱参数信息和与 所述谱参数对应的合成滤波器的系数信息， 并对谱参数或者合成滤波器系 数进行量化、 反量化；
步骤 101 , 进行合成分析搜索， 获得 8kb/s编码速率下的码本参数， 所 述码本参数包括自适应码本参数和固定码本参数， 并进行量化和反量化； 步骤 102 , 根据反量化得到的自适应码本参数和固定码本参数合成 8kb/s速率下的激励信号；
步骤 103 ,利用计算出来的核心层激励信号激励 8kb/s速率反量化后的 合成滤波器， 输出窄带信号分量的重建信号， 应用该重建信号信息更新 8 kb / s速率对应的合成滤波器的状态信息；
步骤 104,应用更新后的 8kb/s速率对应的合成滤波器的状态信息更新 12kb/s对应的合成滤波器的状态信息。
将更新后的 8kb/s速率对应的合成滤波器的状态覆盖 12kb/s对应的合 成滤波器状态即可， 或者在步骤 104中直接使用 8kb/s速率对应的合成滤 波器合成的重建信号更新 12kb/s对应的合成滤波器的状态。
步骤 100中接收到的语音信号是经过预处理过的， 步骤 103中输出窄 带信号分量的重建信号后， 根据该重建信号和经过预处理的语音信号得到 残差信息，将该残差信息经过知觉加权处理后，将残差信息返回给步骤 101 进行合成分析搜索。 因此上述合成分析搜索过程是一个闭环过程。 表 1为 使用的全速率编码的帧结构 20ms帧长比特分配表。
表 1
层 1 核心层 （窄带嵌入式 CELP , 8kb/s)
10 ms f rame 1 10 ms f rame 2 总计 线谱对 (LSP) 18 18 36 subf ramel subf rame2 subf ramel subf rame2
自适应码本延迟 8 5 8 5 26 基音延迟奇偶校验 1 1 2 固定码本索引 13 13 13 13 52 固定码本符号 4 4 4 4 16 码本增益（第一级） 3 3 3 3 12 码本增益（第二级） 4 4 4 4 16
8kb/s核心层总计 160
层 2 窄带增强层 （窄带嵌入式 CELP, 12kb/s) 第二级固定码本索引 13 13 13 13 52 第二级固定码本符号 4 4 4 4 16 第二级固定码本增益 3 2 3 2 10 纠错位 (分类信息） 1 1 2
12kb/s增强层总计 80
层 3 宽带增强层 （TDBWE, 14kb/s)
时域包络均值 5 5 时域包络分裂矢量 7+7 14 频域包络分裂矢量 5+5+4 14 纠错位 (相位信息） 7 7
14kb/s增强层总计 40
层 4至层 12 宽带增强层 （ TDAC, 16kb/s及以上） 纠错位（能量信息） 5 5
MDCT归一化因子 4 4 高带谱包络 nbitS-HB nbits— HB 低带谱包络 nbitS-LB nbits -LB 精细结构 nbitS-VQ = 351 - nbits.HB - nbits— LB nbitS-VQ
16-32kb/s增强层总计 360
总计 640
图 8为本发明合成滤波器状态更新方法实施例二流程图， 当编码速率 从原来的 8kb/s变为 12kb/s或者更高时， 本实施例以编码速率变为 32kb/s 为例介绍编码过程， 如图 8所示包括如下步骤：
步骤 200, 对接收到的语音信号进行 LPC分析， 获得谱参数信息和与 所述谱参数对应的合成滤波器的系数信息， 并对谱参数或者合成滤波器系 数进行量化、 反量化；
步骤 201, 进行合成分析搜索， 获得核心层的码本参数， 包括自适应 码本参数和固定码本参数， 并进行量化、 反量化； 步骤 202, 根据反量化得到的自适应码本参数和固定码本参数合成 8kb/s速率下的激励信号；
步骤 203, 用计算出的核心层激励信号激励 8kb/s的合成滤波器， 并 更新该合成滤波器的状态信息； 步骤 204, 计算窄带增强层的固定码本参数并进行量化、 反量化， 利 用反量化的固定码本参数合成增强的激励信号； 步骤 205 , 用增强的激励信号激励 12kb/s的合成滤波器， 输出窄带信 号分量的重建信号， 并更新该合成滤波器的状态信息； 步骤 206, 应用更新后的 12kb/s的合成滤波器状态更新 8kb/s的合成 滤波器状态； 将更新后的 12kb/s速率对应的合成滤波器的状态覆盖 8kb/s对应的合 成滤波器状态即可，或者在步骤 206中直接使用 12kb/s速率对应的合成滤 波器合成的重建信号更新 8kb/s对应的合成滤波器的状态。
步骤 207 , 应用 TDBWE编码器编码 14kb/s的码流；
步骤 208 , 利用步骤 200中接收到的信号与步骤 205中计算出的重建 信号之间的差值信号， 与高带信号分量一起进行 TDAC编码。
由于解码端仅使用一个合成滤波器并且进行连续的更新， 因此在编码 端进行完步骤 206的操作之后， 保证了步骤 205中重建的窄带信号分量与 解码端重建的窄带信号分量的一致性， 最终会保证解码端重建信号的正确 性。
由上述实施例可知， 在编码时允许每个编码速率下使用独立的合成滤 波器， 并在每帧编码完成后， 不但对当前编码速率所对应的合成滤波器的 状态信息进行更新， 而且还要对其它编码速率对应的合成滤波器的状态信 息进行同步更新， 使得编码端对应于不同编码速率的合成滤波器的状态始 终保持同步， 保证了即使发生编码速率切换时编解码端重建信号的一致 性， 提高解码端重建信号的质量。
本发明合成滤波器状态更新方法实施例三釆用 DTX/CNG技术， 使用 的全速率语音帧的帧结构如表 1所示， 使用的全速率噪音帧的帧结构如表 3所示。 在该实施例中， 对语音帧进行编码时， 釆用与上述实施例中相同 的处理方法对编码速率分别 12kb/s和 8kb/s对应的合成滤波器的状态信息 进行相互更新即可。 若出现噪音帧与语音帧之间的切换的情况， 并且对语 音帧用高于 12kb/s的编码速率进行编码时，由于在对噪音帧信息进行编码 时仅釆用了 8kb/s的合成滤波器进行合成滤波， 为避免出现编解码端重建 窄带信号分量不同步的情况， 在编码器重建噪音信号时， 除了对所使用的 8kb/s的合成滤波器的状态信息进行更新之外，还要应用更新后的 8kb/s的 合成滤波器的状态信息更新 12kb/s的合成滤波器的状态信息，保证编码端 合成滤波器状态的同步性， 进而保证了编解码端重建的窄带信号分量的同 步性。
表 2 参数描述 比特分配 分层结构
LSF参数量化器索引 1 第一级 LSF量化矢量 5
窄带核心层
第二级 LSF量化矢量 4 能量参数量化值 5 能量参数第二级量化值 2
窄带增强层
第三级 LSF量化矢量 4 宽带分量时域包络 6 宽带分量频域包络矢量 1 6
見 Tff核心层
宽带分量频域包络矢量 2 6 宽带分量频域包络矢量 3 6 虽然在上述实施例中涉及的 CELP 编码器仅介绍了支持 8kb/s 和 12kb/s两种编码速率的情况， 但是本发明实施例提供的合成滤波器状态更 新方法并不局限于两种编码速率之间的切换， 也可以应用于更多的 CELP 编码速率的情况， 只要将各个编码速率下的合成滤波器的状态信息进行同 步处理即可。
本领域普通技术人员可以理解： 实现上述方法实施例的全部或部分步 骤可以通过程序指令相关的硬件来完成， 前述的程序可以存储于一计算机 可读取存储介质中， 该程序在执行时， 执行包括上述方法实施例的步骤； 而前述的存储介质包括： ROM、 RAM, 磁碟或者光盘等各种可以存储程 序代码的介质。
合成滤波器状态更新装置实施例
该合成滤波器状态更新装置包括数个合成滤波器， 还包括状态更新模 块， 用于在利用第一编码速率的激励信号对所述第一编码速率对应的合成 滤波器进行激励， 输出重建信号信息后， 更新所述合成滤波器和第二编码 速率对应的合成滤波器的状态信息。
进一步地， 根据更新方式的不同， 状态更新模块可以有不同的组成方 式， 例如可包括第一更新子模块， 用于利用所述重建信号信息， 更新所述 第一编码速率对应的合成滤波器的状态信息； 和第二更新子模块， 用于利 用更新后的第一编码速率对应的合成滤波器的状态信息， 更新所述第二编 码速率对应的合成滤波器的状态信息。 状态更新模块还可以由以下组成方 式， 包括第一更新子模块， 用于利用所述重建信号信息， 更新所述第一编 码速率对应的合成滤波器的状态信息； 和第三更新子模块， 用于利用所述 重建信号信息， 更新所述第二编码速率对应的合成滤波器的状态信息。
图 9为本发明合成滤波器状态更新装置结构示意图， 具体为 G.729.1 中 CELP编码器结构示意图，如图 9所示，对于编码速率为 8kb/s和 12kb/s 的合成滤波器， 还是釆用两个独立的第一合成滤波器 1和第二合成滤波器 2, 并且应用两个独立的第一激励信号合成模块 3 和第二激励信号合成模 块 4分别对对应的合成滤波器进行激励。 根据当前编码速率选择一路合成 滤波器， 在 LPC系数确定模块 5确定完 LPC系数后， 应用所选择的合成 滤波器进行窄带信号分量的重建， 输出重建信号信息， 并应用重建信号通 过状态更新模块 6对当前编码速率如 8kb/s对应的合成滤波器进行状态更 新； 并且还要应用更新后的合成滤波器的状态， 通过状态更新模块 6更新 编码速率为 12kb/s对应合成滤波器的状态，使得第一合成滤波器 1和第二 合成滤波器 2的状态保持同步。
在解码端釆用与现有 G.729.1 中 CELP解码器结构相同的一个合成滤 波器即可。 应用本实施例提供的合成滤波器状态更新装置， 通过应用状态 更新模块同时对编码器中， 对应于不同编码速率的合成滤波器的状态进行 更新， 保证了编码端不同速率下合成滤波器状态的同步， 在编码速率发生 切换时保证编解码端重建信号的一致性， 提高解码端重建信号的质量。
最后应说明的是： 以上实施例仅用以说明本发明的技术方案， 而非对 其限制； 尽管参照前述实施例对本发明进行了详细的说明， 本领域的普通 技术人员应当理解： 其依然可以对前述各实施例所记载的技术方案进行修 改， 或者对其中部分技术特征进行等同替换； 而这些修改或者替换， 并不 使相应技术方案的本质脱离本发明各实施例技术方案的精神和范围。

Claims

权 利 要 求 书
1、 一种合成滤波器状态更新方法， 其特征在于， 包括：
在利用第一编码速率的激励信号对所述第一编码速率对应的合成滤 波器进行激励， 输出重建信号信息后， 更新所述合成滤波器和第二编码速 率对应的合成滤波器的状态信息。
2、 根据权利要求 1 所述的合成滤波器状态更新方法， 其特征在于， 所述更新所述合成滤波器和第二编码速率对应的合成滤波器的状态信息 具体为：
利用所述重建信号信息， 更新所述第一编码速率对应的合成滤波器的 状态信息；
利用更新后的第一编码速率对应的合成滤波器的状态信息， 更新所述 第二编码速率对应的合成滤波器的状态信息。
3、 根据权利要求 1 所述的合成滤波器状态更新方法， 其特征在于， 所述更新所述合成滤波器和第二编码速率对应的合成滤波器的状态信息 具体为：
利用所述重建信号信息， 更新所述第一编码速率和所述第二编码速率 对应的合成滤波器的状态信息。
4、 一种合成滤波器状态更新装置， 包括数个合成滤波器， 其特征在 于还包括状态更新模块， 用于在利用第一编码速率的激励信号对所述第一 编码速率对应的合成滤波器进行激励， 输出重建信号信息后， 更新所述合 成滤波器和第二编码速率对应的合成滤波器的状态信息。
5、 根据权利要求 4所述的合成滤波器状态更新装置， 其特征在于， 所述状态更新模块包括：
第一更新子模块， 用于利用所述重建信号信息， 更新所述第一编码速 率对应的合成滤波器的状态信息；
第二更新子模块， 用于利用更新后的第一编码速率对应的合成滤波器 的状态信息， 更新所述第二编码速率对应的合成滤波器的状态信息。
6、 根据权利要求 4所述的合成滤波器状态更新装置， 其特征在于， 所述状态更新模块包括：
第一更新子模块， 用于利用所述重建信号信息， 更新所述第一编码速 率对应的合成滤波器的状态信息；
第三更新子模块， 用于利用所述重建信号信息， 更新所述第二编码速 率对应的合成滤波器的状态信息。
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