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WO2008141579A1 - Procédé de codage et de décodage de signal audio transitoire - Google Patents

Procédé de codage et de décodage de signal audio transitoire Download PDF

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Publication number
WO2008141579A1
WO2008141579A1 PCT/CN2008/071007 CN2008071007W WO2008141579A1 WO 2008141579 A1 WO2008141579 A1 WO 2008141579A1 CN 2008071007 W CN2008071007 W CN 2008071007W WO 2008141579 A1 WO2008141579 A1 WO 2008141579A1
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WO
WIPO (PCT)
Prior art keywords
transient signal
encoding
processing
signal
audio
Prior art date
Legal status (The legal status 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 status listed.)
Ceased
Application number
PCT/CN2008/071007
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English (en)
Chinese (zh)
Inventor
Heyun Huang
Benhao Zhang
Tan Li
Fuhui Lin
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Spreadtrum Communications Shanghai Co Ltd
Original Assignee
Spreadtrum Communications Shanghai 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.)
Filing date
Publication date
Application filed by Spreadtrum Communications Shanghai Co Ltd filed Critical Spreadtrum Communications Shanghai Co Ltd
Publication of WO2008141579A1 publication Critical patent/WO2008141579A1/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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Classifications

    • 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
    • G10L19/26—Pre-filtering or post-filtering
    • G10L19/265—Pre-filtering, e.g. high frequency emphasis prior to 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
    • G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/0212—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 spectral analysis, e.g. transform vocoders or subband vocoders using orthogonal transformation
    • 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
    • G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/022—Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring
    • G10L19/025—Detection of transients or attacks for time/frequency resolution switching

Definitions

  • the present invention relates to the field of digital mobile communications, and in particular to a method for encoding and decoding audio transient signals. Background technique
  • a transient signal is a special kind of audio signal, which is mostly present in an audio sequence with a percussion instrument.
  • a signal generated by a continuous tapping drum can be called a transient signal.
  • MDCT Modified Discrete Cosine Transform
  • a pre-echo phenomenon will occur. This is due to the quantization noise caused by insufficient quantization bits.
  • the quantization noise is uniformly spread throughout the time domain, and the signal before the occurrence of the transient signal is occupied by the quantization noise, which in turn produces a pre-echo phenomenon.
  • Pre-echo phenomenon is a serious distortion, and the human ear is very sensitive to such distortion.
  • the technical problem to be solved by the present invention is to provide a method for encoding and decoding an audio transient signal, which can effectively reduce the computational complexity of the codec.
  • the encoding and decoding method of the audio transient signal of the present invention is implemented by adopting the following technical solution, and processing the time domain sampling point of the input frame, ie adding the transient state, before performing transform coding on the transient signal Signal processing;
  • the transient signal processing is located at any position before the time-frequency transform at the encoder side and at any position after the corresponding decoder side performs the frequency-time transform.
  • the method of the present invention is an improvement over the prior art gain correction method. Since the gain correction method corrects the gain of the entire transient signal frame, the number of bits required for accurate correction is quite high. In fact, since the transient signal is only present in a small portion of the transient signal frame. Therefore, it is only necessary to perform gain correction on a part of the time domain sampling points instead of all the time domain sampling points. By adopting the invention, the number of bits to be quantized can be reduced, and the computational complexity of the codec side can be reduced at the same time. BRIEF abstract
  • Figure 1 is a flow chart showing the control of the encoding end in the method of the present invention
  • 2 is a control flow chart of a decoding end in the method of the present invention
  • FIG. 3 is a block diagram of a transient signal processing of an encoding end in the method of the present invention.
  • Transient signal processing can be performed anywhere the frequency-time transform is performed at any position before the encoder performs the time-frequency transform and the corresponding decoder.
  • the time domain processing includes the processing steps of the input sampling points, such as an analysis filter.
  • the time-frequency transform (or frequency-time transform) includes a transform code (de) code followed by some frequency (time) domain processing and quantization steps.
  • de transform code
  • time time domain processing
  • the input frame is a transient frame.
  • Any known transient signal detection method eg, perceptual entropy
  • the average energy of the entire frame ⁇ : . ⁇ Compare the average ratio of the average energy of the entire frame to the energy of each segment with the set threshold. When the maximum ratio is greater than the threshold ⁇ , the input frame is a transient frame, otherwise it is steady state. frame. which is:
  • Max->T The essential reason for the pre-echo signal generation is the time period before the quantization noise spreads to the transient signal, and the anterior masking of the hearing is not enough, so the quantization noise is clearly felt to the human ear, resulting in the final seriousness. distortion.
  • the transient signal processing method When the input frame is detected as a transient signal, the transient signal processing method is turned on.
  • transient signal processing method is turned on.
  • a typical processing method is the long and short window switching method. It switches it to when the transient signal appears The short window increases the time resolution of the signal, thereby suppressing the pre-echo phenomenon. This method has been applied in many standards and proved to be a good method.
  • TNS Temporal Noise Shaping
  • the Bit Pool method is also an effective method.
  • the temporary signal is coded and decoded by using the previously saved bits.
  • the quantization noise is reduced.
  • the pre-echo phenomenon can be alleviated to some extent.
  • the Gain Modification method is also a classic algorithm. It calculates some gain and shape parameters based on the time domain sampling points of the transient signal, and uses these parameters to correct the time domain sampling points to achieve the suppression of the pre-echo phenomenon.
  • the method of processing transient signals of the present invention is an improvement over the gain correction method. Since the gain correction method corrects the gain of the entire transient signal frame, the number of bits required for accurate correction is quite high. In fact, since the transient signal is only present in a small portion of the transient signal frame. Therefore, it is only necessary to perform gain correction on a part of the time domain sampling points instead of all the time domain sampling points. This can reduce the number of bits that need to be quantized, and at the same time reduce the computational complexity of the codec.
  • sample points of the input frame are: ⁇ 2 '...' ⁇ (denoted as set C) , where N is the input frame length.
  • scaling processing i.e., gain correction
  • ⁇ ⁇ , , , . , .,
  • Multiplicative parameter ⁇ ' ' ⁇ , ⁇ can be obtained by any method. It can be pre-set, or it can be adaptively calculated based on input sample points and other parameters (such as sample rate, code rate, etc.).
  • the data of the set A After the data of the set A is processed, the data of the set A and the data of the set B are rearranged in the original order of the C set to obtain a transient signal that has been scaled and processed, and is used as an input of the time-frequency transform coding.
  • the implementation process can be seen in Figure 3.
  • a transient signal time domain recovery processing module is applied to recover the time domain signal.
  • the time domain recovery signal sampling points can be obtained:
  • the inverse scaling process is performed according to the collection ⁇ and the collection B generated by the encoding end:
  • the decoding end multiplicative parameter ⁇ can be obtained by any method. It can be pre-set, or it can be obtained by any codec method by multiplying the multiplier parameters of the encoder. According to the above process, the output X1 '''''''' XAr of the transient signal processing is generated. See Figure 4 for the recovery process.

Landscapes

  • 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

Un procédé de codage et de décodage de signal audio transitoire consiste à traiter des points d'échantillonnage dans le domaine temporel de trames d'entrée, à savoir à ajouter le traitement d'un signal transitoire (2) avant la transformation du signal transitoire. Le traitement du signal transitoire est situé dans n'importe quelle position avant la transformation temps-fréquence (3) au niveau d'une extrémité de codeur et dans n'importe quelle position après la transformation fréquence-temps à une extrémité de décodeur correspondante.
PCT/CN2008/071007 2007-05-17 2008-05-19 Procédé de codage et de décodage de signal audio transitoire Ceased WO2008141579A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CNA200710040779XA CN101308656A (zh) 2007-05-17 2007-05-17 音频暂态信号的编解码方法
CN200710040779.X 2007-05-17

Publications (1)

Publication Number Publication Date
WO2008141579A1 true WO2008141579A1 (fr) 2008-11-27

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PCT/CN2008/071007 Ceased WO2008141579A1 (fr) 2007-05-17 2008-05-19 Procédé de codage et de décodage de signal audio transitoire

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CN (1) CN101308656A (fr)
WO (1) WO2008141579A1 (fr)

Families Citing this family (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CA2777182C (fr) * 2009-10-09 2016-11-08 Dts, Inc. Amelioration adaptative de plage dynamique d'enregistrements audio
CN101694773B (zh) * 2009-10-29 2011-06-22 北京理工大学 一种基于tda域的自适应窗切换方法
FR2969805A1 (fr) * 2010-12-23 2012-06-29 France Telecom Codage bas retard alternant codage predictif et codage par transformee
CN103295577B (zh) * 2013-05-27 2015-09-02 深圳广晟信源技术有限公司 用于音频信号编码的分析窗切换方法和装置
CN104992711B (zh) * 2015-05-27 2018-06-15 东南大学 一种基于移动终端的局域网集群双工语音通信方法

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5825320A (en) * 1996-03-19 1998-10-20 Sony Corporation Gain control method for audio encoding device
CN1552060A (zh) * 2001-05-10 2004-12-01 ʵ 通过降低前噪声改善低比特速率音频编码系统的瞬时性能
WO2007028280A1 (fr) * 2005-09-08 2007-03-15 Beijing E-World Technology Co., Ltd. Codeur et decodeur pour commande de pre echo et son procede

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5825320A (en) * 1996-03-19 1998-10-20 Sony Corporation Gain control method for audio encoding device
CN1552060A (zh) * 2001-05-10 2004-12-01 ʵ 通过降低前噪声改善低比特速率音频编码系统的瞬时性能
WO2007028280A1 (fr) * 2005-09-08 2007-03-15 Beijing E-World Technology Co., Ltd. Codeur et decodeur pour commande de pre echo et son procede

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
HE B. ET AL.: "A Window Switching Method of Audio Coding in Low Code Rate", ELECTROACOUSTIC TECHNIQUE, November 2001 (2001-11-01), pages 3 - 6 *
JIANG Y. ET AL.: "A Generation Theory of Pre-echoes in Perceptual Audio Coding and a Method for Reducing pre-echoes", ELECTROACOUSTIC TECHNIQUE, November 2000 (2000-11-01), pages 15 - 18 *

Also Published As

Publication number Publication date
CN101308656A (zh) 2008-11-19

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