WO2008110109A1 - Procédé et appareil pour le lissage de gains dans un décodeur vocal - Google Patents

Procédé et appareil pour le lissage de gains dans un décodeur vocal Download PDF

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Publication number
WO2008110109A1
WO2008110109A1 PCT/CN2008/070458 CN2008070458W WO2008110109A1 WO 2008110109 A1 WO2008110109 A1 WO 2008110109A1 CN 2008070458 W CN2008070458 W CN 2008070458W WO 2008110109 A1 WO2008110109 A1 WO 2008110109A1
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WIPO (PCT)
Prior art keywords
frame
speech
fixed codebook
current
codebook gain
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PCT/CN2008/070458
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English (en)
Chinese (zh)
Inventor
Jianfeng Xu
Lijing Xu
Qing Zhang
Wei Li
Shenghu Sang
Zhengzhong Du
Chen Hu
Original Assignee
Huawei Technologies Co., Ltd.
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
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Application filed by Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Publication of WO2008110109A1 publication Critical patent/WO2008110109A1/fr

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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
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/083—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being an excitation gain

Definitions

  • the present invention relates to the field of speech decoding technology, and more particularly to a method and apparatus for performing gain smoothing in a speech decoder. Background technique
  • an encoder encodes an input voice signal, and then transmits the encoded bit stream through a communication channel; the decoder performs a bit stream received from the communication channel. After decoding, it is synthesized into a speech signal.
  • An encoder that encodes a speech signal is hereinafter referred to as a speech coder.
  • the coding principle commonly used in speech encoders is Algebraic Code Excited Linear Prediction (ACELP).
  • ACELP Algebraic Code Excited Linear Prediction
  • Such encoders include G.729, EVRC, AMR, AMR-WB, AMR-WB+ and so on.
  • G.729 is the voice coding standard of the International Telecommunication Union (ITU-T);
  • EVRC is the speech coding standard of the 3rd Generation Partnership Project 2 (3GPP-2);
  • AMR, AMR-WB AMR-WB+ is the speech coding standard of the 3rd Generation Partnership Project (3GPP).
  • the code streams generated by the ACELP-based speech coder are all in units of speech frames, and some divide the frame into sub-frames.
  • AMR is in units of sub-frames.
  • the speech coder at the transmitting end encodes it as a set of parameters. These parameters are typically quantized and transmitted.
  • the decoder at the receiving end recomposes these parameters into a voice signal, which is usually PCM format data.
  • the parameters of the speech frame generated by the ACELP-based speech coder generally include speech parameters, adaptive codebook parameters, algebraic codebook parameters, adaptive codebook gains, and algebraic codebook gains.
  • some post-processing such as fixed codebook gain smoothing and enhanced periodic processing, is generally performed to improve the synthesized speech quality.
  • the purpose of the fixed codebook gain smoothing is to avoid steady state language. The energy of the sound does not fluctuate naturally.
  • the speech decoder uses two methods to perform gain smoothing on the fixed code.
  • One is to smooth the fixed codebook based on the stability of the Linear Spectral Pair (LSP), and the other is to perform gain smoothing on the fixed codebook based on the stability and voiced characteristics of the speech.
  • LSP Linear Spectral Pair
  • Ev is the energy of the adaptive codebook and Ec is the energy of the fixed codebook.
  • the second disadvantage of the gain smoothing method is that it is necessary to calculate the stability factor and the voiced factor, and the algorithm complexity is large.
  • the speech encoder performs gain smoothing on the fixed code, since the average codebook average gain of the past multiple subframes needs to be recorded, or the stability factor and the voiced sound factor need to be calculated, the speech coding process is performed. very complicated. Summary of the invention
  • the main object of the present invention is to provide a method and apparatus for performing gain smoothing in a speech decoder for simplifying the smoothing of gain in speech coding.
  • the fixed codebook gain of the current speech frame is smoothed by the initialization fixed corrected codebook gain and the smoothing factor corresponding to the speech frame of the state.
  • An apparatus for performing gain smoothing in voice decoding includes: a voice parameter change factor acquiring unit, configured to acquire a voice parameter change factor of a current frame; a fixed codebook gain initialization correction unit, configured to perform initialization correction on a fixed codebook gain of the current speech frame;
  • a voice frame state determining unit configured to determine a state of the current voice frame according to the obtained voice parameter change factor of the current frame
  • the fixed codebook gain smoothing unit is configured to smooth the fixed codebook gain of the current speech frame according to the initialization fixed corrected codebook gain and the smoothing factor corresponding to the speech frame of the state.
  • the embodiment of the present invention only needs to record the corrected fixed codebook gain of the previous frame, and does not need to record the fixed codebook average gain of consecutive multiple frames, so the storage is simple. Moreover, the embodiment of the present invention only needs to use a speech parameter variation factor of the current speech frame to achieve smoothing of the gain, and does not need to simultaneously calculate the stability factor and the voiced sound factor, so the algorithm complexity is also reduced.
  • FIG. 1 is a schematic diagram of a voice communication system in the prior art
  • FIG. 2 is a schematic flow chart of smoothing a fixed codebook gain according to an embodiment of the present invention
  • FIG. 3 is a schematic flowchart of smoothing a fixed codebook gain based on a spectral parameter change factor according to another embodiment of the present invention
  • FIG. 4 is a schematic flow chart of smoothing a fixed codebook gain based on a spectral parameter variation factor according to still another embodiment of the present invention.
  • FIG. 5 is a schematic flowchart of smoothing a fixed codebook gain based on a pitch delay parameter variation factor according to still another embodiment of the present invention.
  • FIG. 6 is a schematic structural diagram of an apparatus for performing gain smoothing in a speech decoder according to an embodiment of the present invention
  • FIG. 7 is a schematic structural diagram of a specific embodiment of the apparatus of the present invention. detailed description
  • An embodiment of the present invention is to calculate a voice parameter of the current voice frame in a voice communication system.
  • a change factor initializing and correcting a fixed codebook gain of the current speech frame; determining a state of the current speech frame according to the speech parameter variation factor; using the initialization fixed corrected codebook gain and a speech frame for the state
  • the smoothing factor is set to smooth the fixed codebook gain of the current speech frame.
  • the speech parameter change factor of the current speech frame can be calculated by using the speech parameters of the current frame and the speech parameters of the previous frame.
  • the speech parameters can be spectral parameters, pitch delay parameters, or voiced factors.
  • the smoothing factor can be calculated using a certain formula or based on the simulation results.
  • Step 201 Calculating a voice parameter change factor of the current voice frame by using a voice parameter of a current frame and a voice parameter of a previous frame, And performing initialization correction on the fixed codebook gain of the current speech frame; if the speech parameter change factor is a spectral parameter change factor, step 201 may use the spectral parameters of the current frame and the spectral parameters of the previous frame to calculate the current frame. Speech parameter change factor. If the speech parameter variation factor is the pitch delay parameter variation factor, step 201 can be calculated using the pitch delay parameter of the current frame and the pitch delay parameter of the previous frame.
  • the initial codebook gain needs to be set to:
  • the previous speech frame is initialized.
  • the initial code correction needs to be set to: The previous speech frame is initialized and corrected.
  • Step 202 Determine a state of the current voice frame according to the voice parameter change factor.
  • the voice frame may be divided into several states according to the voice parameter change factor, and the correspondence between the state of each voice frame and the voice parameter change range may be set. Then, the state of the current voice frame may be determined in step 202: And determining a range of the voice parameter in which the voice frame parameter change factor is located; and obtaining a state of the current voice frame corresponding to the voice parameter change range according to the correspondence between the state of the voice frame and the voice parameter change range.
  • Step 203 Smooth the fixed codebook gain of the current speech frame by using the initialization fixed codebook gain and the smoothing factor corresponding to the speech frame of the state.
  • the formula for smoothing is ⁇ + ⁇ - ⁇ . ⁇ , where the smoothing factor set for the speech frame of this state is the fixed codebook gain of the current speech frame.
  • another embodiment of the present invention smoothes the fixed codebook gain based on the spectral parameter variation factor as follows:
  • Step 301 Calculate a spectral parameter variation factor of the current frame by using a spectral parameter of the current frame and a spectral parameter of the previous frame. Calculated as follows:
  • spectral parameter of the current frame which is the spectral parameter of the previous frame
  • / is a function of 5 and .
  • the spectral parameters may be ISF or ISP or LSP or LSF or LPC.
  • Different speech codecs may use one or more of ISF, ISP, LSP, LSF, LPC to represent the short-term correlation of the speech signal.
  • Step 302 Initialize and correct the fixed codebook gain of the current speech frame.
  • the fixed codebook gain of the previous speech frame may be initialized and corrected by using the fixed codebook gain after initialization correction.
  • Step 303 Determine a state of the current voice frame according to the voice parameter change factor.
  • the speech frame is divided into n+1 states according to the speech parameter change factor, n is a natural number, ... t n is a speech frame state threshold, therefore, the speech parameter variation range can be set to be smaller than or greater than ⁇ is less than t 2 . and greater than ⁇ a total of n+1 speech parameter variation ranges, each speech parameter variation range corresponds to a state of a speech frame;
  • the voice parameter variation range of the current voice frame can be determined according to the voice change factor, and then the state of the current voice frame corresponding to the voice parameter variation range is determined.
  • Step 304 Smooth the fixed codebook gain of the current speech frame by using the initialization fixed codebook gain and the smoothing factor corresponding to the state.
  • the value of the smoothing factor may be determined based on the simulation result. It is also necessary to preset the correspondence between the speech parameter variation range and the smoothing factor.
  • the fixed codebook gain of the current speech frame is a spectral parameter change factor.
  • the smoothing factor in the state is when the speech parameter variation factor is greater than or equal to ⁇
  • the smoothing factor in the state is; when the speech parameter change factor is greater than or equal to t m4 and less than t m , it is the mth state, and the smoothing factor in the state is the speech parameter.
  • the variation factor is greater than or equal to ⁇ , it is the nth state, and the smoothing factor in this state is & ,
  • the smoothing process is ⁇ + ⁇ - ⁇ , where the smoothing factor set for the speech frame of the state is the fixed codebook gain of the current speech frame.
  • a method for smoothing a fixed codebook gain based on a spectral parameter variation factor includes the following steps:
  • Step 401 Calculate a spectral parameter change factor s - di ff by using a spectral parameter of the current frame and a spectral parameter of a previous frame, and the spectral parameter change factor may be a change factor of LSF, ISF, LPC, ISP, LSP, etc., and the calculation formula may be for:
  • the s-scale is a normalization factor and can be a constant, for example, it can take the value 40000.
  • Step 402 Using the fixed codebook gain after initialization correction of the previous speech frame, or the ratio of the fixed codebook gain to the gain scaling factor, or the product of the fixed codebook gain and the gain scaling factor, as the fixed current frame is fixed. Codebook gain.
  • the fixed codebook gain after the initial correction of the current speech frame is: the fixed code after the initial speech frame is initialized and corrected.
  • the fixed codebook gain after the initial correction of the current speech frame is: The codebook gain, and the minimum of the product of the fixed codebook gain and the gain scaling factor.
  • Step 403 Determine the state of the current voice frame according to the voice parameter change factor.
  • a speech frame state threshold may be preset according to the simulation result, and the speech frames are classified into two types according to the steady state and the non-steady state.
  • L is greater than the speech frame state threshold, it indicates that the spectral parameter is in an unsteady state.
  • L is less than or equal to the speech frame state threshold, the spectral parameter is in a steady state. Therefore, it can be stable and unsteady according to the simulation result.
  • Set a fixed codebook smoothing factor, and the smoothing factor at steady state is smaller than the unsteady smoothing factor.
  • Step 404 Smooth the fixed codebook gain of the current speech frame by using the initialized fixed codebook gain and a smoothing factor set for the speech frame of the state.
  • ⁇ c S 2 - g 0 + ⁇ -S 2 ) ⁇ c smooth the fixed codebook gain of the speech frame;
  • the fixed codebook gain of the current speech frame is a speech frame state threshold, which may be a constant, such as 0.58. Greater than t/?r indicates that the spectral parameters are in an unsteady state, and less than t/?r indicates that the spectral parameters are in a steady state. And are corresponding to two different types of fixed codebook gain smoothing factors, which are constants, for example, 0.17 and 0.83 respectively.
  • a specific process for smoothing the fixed codebook gain based on the pitch delay parameter variation factor in another embodiment of the present invention is as follows:
  • Step 501 Calculate a pitch delay parameter change factor - according to a pitch delay parameter of the current frame and a pitch delay parameter of the previous frame.
  • the formula can be as follows:
  • Step 502 Perform initial correction on the fixed codebook gain. Details as follows:
  • Step 503 Determine the state of the current speech frame according to the pitch delay parameter change factor ⁇ -. For the determination method, refer to step 303.
  • Step 504 Perform gain smoothing on the fixed codebook gain of the current speech frame according to the smoothing factor corresponding to the state.
  • n is a natural number, which is the fixed codebook gain of the current speech frame.
  • the apparatus for performing gain smoothing in a speech decoder includes: a speech parameter change factor acquisition unit 61, a fixed codebook gain initialization correction unit 62, a speech frame state determination unit 63, and a fixed codebook gain. Smoothing unit 64.
  • the voice parameter change factor obtaining unit 61 is configured to acquire a change factor of the voice parameter of the current frame.
  • the fixed codebook gain initialization correction unit 62 is configured to perform initialization correction on the fixed codebook gain of the current speech frame.
  • the speech frame state determining unit 63 is configured to determine the current speech according to the obtained speech parameter variation factor of the current frame.
  • the state of the frame; the fixed codebook gain smoothing unit 64 is configured to smooth the fixed codebook gain of the current speech frame according to the initialization of the fixed fixed codebook gain and the smoothing factor corresponding to the speech frame of the state.
  • the voice parameter change factor obtaining unit 61 may include: a first voice parameter acquiring unit 71, a second voice parameter acquiring unit 72, and a voice parameter change factor calculating unit 73, wherein the first voice parameter acquiring unit 71, The second speech parameter acquisition unit 72 is configured to acquire the speech parameter of the previous frame, and the speech parameter change factor calculation unit 73 is configured to use the speech parameter of the current frame and the speech of the previous frame.
  • the parameter calculates a speech parameter of the current frame, ⁇ factor.
  • the voice frame state determining unit 63 may include: a storage unit 74 and a voice frame state analyzing unit
  • the storage unit 74 is configured to save a correspondence between a state of the voice frame and a range of the voice parameter change; the voice frame state parsing unit 75 is configured to determine a range of the voice parameter in which the obtained voice frame parameter change factor is located; The correspondence stored by the unit 74 obtains the state of the current speech frame corresponding to the range of the speech parameter variation.
  • the fixed codebook gain smoothing unit 64 may include: a smoothing factor storage unit 76, a smoothing factor acquisition unit 77, and a smoothing processing unit 78.
  • the smoothing factor storage unit 76 is configured to store a correspondence between a state of the voice frame and a smoothing factor.
  • the smoothing factor obtaining unit 77 is configured to: according to the state of the current voice frame, the state of the voice frame stored by the smoothing factor storage unit 76. Corresponding relationship with the smoothing factor, obtaining a smoothing factor corresponding to the speech frame of the state; the smoothing processing unit 78 is configured to smooth the fixed codebook gain of the current speech frame according to ⁇ + ⁇ - ⁇
  • the smoothing factor set for the speech frame of the state is the fixed codebook gain of the current speech frame.
  • the fixed codebook gain initialization correction unit 62 includes: a comparison unit 79 and a correction processing unit 70.
  • the comparing unit 79 is configured to determine whether the fixed codebook gain of the current frame is greater than the fixed codebook gain after the initial correction of the previous speech frame.
  • the correction processing unit 70 is configured to use the fixed codebook gain of the current frame to be greater than the previous speech.
  • the fixed codebook gain of the current frame is set to: the fixed codebook gain after the initial correction of the previous speech frame, and the ratio of the fixed codebook gain to the gain scaling factor. Maximum value;
  • the fixed codebook gain of the current frame is less than or equal to the fixed codebook gain after the initial correction of the previous speech frame, the fixed codebook gain of the current frame is set to: The fixed code after the initial speech frame is initialized and corrected. The gain, and the minimum of the product of the fixed codebook gain and the gain scaling factor.
  • the fixed codebook gain of the current voice frame is initialized and corrected by calculating a voice parameter change factor of the current voice frame; determining a state of the current voice frame according to the voice parameter change factor; and using the initialized fixed code
  • the gain and the smoothing factor set for the speech frame of the state smooth the fixed codebook gain of the current speech frame.
  • the smoothing factor can be calculated by using a certain formula, and can also be obtained according to the simulation result.

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

L'invention concerne un procédé pour le lissage de gains dans un décodeur vocal, lequel procédé comprend le calcul des facteurs de variation de paramètre de voix d'une trame de voix courante, et la modification initiale d'un gain de livre de code fixe de ladite trame de voix courante (201); la détermination de l'état de la trame de voix courante sur la base desdits facteurs de variation de paramètre de voix (202); le lissage du gain de livre de code fixe de la trame de voix courante à l'aide du gain de livre de code fixe qui est initialement modifié et d'un facteur de lissage qui est défini pour l'état de la trame de voix (203).
PCT/CN2008/070458 2007-03-12 2008-03-10 Procédé et appareil pour le lissage de gains dans un décodeur vocal WO2008110109A1 (fr)

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CN200710088039.3 2007-03-12
CN 200710088039 CN101266798B (zh) 2007-03-12 2007-03-12 一种在语音解码器中进行增益平滑的方法及装置

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CN106887233B (zh) * 2015-12-15 2020-01-24 广州酷狗计算机科技有限公司 音频数据处理方法及系统
CN113205824B (zh) * 2021-04-30 2022-11-11 紫光展锐(重庆)科技有限公司 声音信号处理方法、装置、存储介质、芯片及相关设备

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