WO2010130225A1 - 一种音频解码方法和音频解码器 - Google Patents

一种音频解码方法和音频解码器 Download PDF

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WO2010130225A1
WO2010130225A1 PCT/CN2010/072781 CN2010072781W WO2010130225A1 WO 2010130225 A1 WO2010130225 A1 WO 2010130225A1 CN 2010072781 W CN2010072781 W CN 2010072781W WO 2010130225 A1 WO2010130225 A1 WO 2010130225A1
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frequency domain
mono
domain signal
decoding
energy
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French (fr)
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张琦
张立斌
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Huawei Technologies Co Ltd
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Huawei Technologies Co Ltd
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Priority to JP2012510106A priority Critical patent/JP5418930B2/ja
Priority to KR1020117028589A priority patent/KR101343898B1/ko
Priority to EP10774566.3A priority patent/EP2431971B1/en
Publication of WO2010130225A1 publication Critical patent/WO2010130225A1/zh
Priority to US13/296,001 priority patent/US8620673B2/en
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/16Vocoder architecture
    • G10L19/18Vocoders using multiple modes
    • G10L19/24Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/008Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/02Speech 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
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04HBROADCAST COMMUNICATION
    • H04H20/00Arrangements for broadcast or for distribution combined with broadcast
    • H04H20/86Arrangements characterised by the broadcast information itself
    • H04H20/88Stereophonic broadcast systems
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S1/00Two-channel systems
    • H04S1/002Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S3/00Systems employing more than two channels, e.g. quadraphonic
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04HBROADCAST COMMUNICATION
    • H04H20/00Arrangements for broadcast or for distribution combined with broadcast
    • H04H20/86Arrangements characterised by the broadcast information itself
    • H04H20/95Arrangements characterised by the broadcast information itself characterised by a specific format, e.g. an encoded audio stream
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04HBROADCAST COMMUNICATION
    • H04H40/00Arrangements specially adapted for receiving broadcast information
    • H04H40/18Arrangements characterised by circuits or components specially adapted for receiving
    • H04H40/27Arrangements characterised by circuits or components specially adapted for receiving specially adapted for broadcast systems covered by groups H04H20/53 - H04H20/95
    • H04H40/36Arrangements characterised by circuits or components specially adapted for receiving specially adapted for broadcast systems covered by groups H04H20/53 - H04H20/95 specially adapted for stereophonic broadcast receiving

Definitions

  • the present invention relates to the field of multi-channel audio codec technology, and in particular to an audio decoding method and an audio decoder. Background technique
  • multi-channel audio signals have a wide range of application scenarios, such as teleconferencing, games, etc., so the encoding and decoding of multi-channel audio signals is also receiving more and more attention.
  • Traditional encoders based on waveform coding such as MPEG-II (Moving Picture Experts Group II), MP3 (Moving Picture Experts Group Audio Layer III) and AAC ( Advanced Audio Coding, when encoding multi-channel signals, encodes each channel independently. Although this method can recover a multi-channel signal well, the required bandwidth and code rate are several times that of the mono signal.
  • the more popular stereo or multi-channel coding technology is parametric stereo coding, which can reconstruct a multi-channel signal with the same auditory experience and original signal with a small bandwidth.
  • the basic method is: at the encoding end, the multi-channel signal is down-mixed into a mono signal, and the signal is independently encoded, and the channel parameters between the channels are extracted, and the parameters are encoded.
  • the downmixed mono signal is decoded first, then the channel parameters between the channels are decoded, and finally the multichannels are synthesized together with the downmixed mono signals using these channel parameters. signal.
  • Typical parametric stereo coding techniques such as PS (Variable Stereo), are widely used.
  • the channel parameters commonly used to describe the relationship between channels in parametric stereo coding are
  • ITD Inter-channel Time Difference
  • ILD Inter-channel Level Difference
  • ICC Inter-Channel Coherence
  • the embodiment of the invention provides an audio decoding method and an audio decoder, which can make the codec end process the signals consistent and improve the quality of the decoded stereo signal.
  • An audio decoding method includes:
  • the left and right channel frequency domain signals are reconstructed in the second sub-band region using the mono-decoded frequency domain signal that is not energy-adjusted.
  • An audio decoder comprising: a determining unit, a processing unit and a first reconstructing unit, wherein: the determining unit is configured to determine whether the code stream to be decoded is a mono coding layer and a stereo first enhancement layer code stream If yes, triggering the first reconstruction unit;
  • the processing unit is configured to decode the mono coding layer to obtain mono decoding Frequency domain signal
  • the first reconstruction unit is configured to reconstruct the left and right channel frequency domain signals by using the energy-adjusted mono decoding frequency domain signal in the first sub-band region; and adopting the processing unit in the second sub-band region
  • the decoded unchannelized frequency modulated frequency domain signal obtained by the decoding reconstructs the left and right channel frequency domain signals.
  • the embodiment of the present invention determines a mono signal type used in reconstructing a mono signal in a decoding process according to a code stream state to be decoded, wherein the code stream to be decoded is determined to be a mono coding layer and a stereo.
  • the energy-adjusted mono decoding frequency domain signal is used to reconstruct the left and right channel frequency domain signals in the first sub-band region; the energy-adjusted single is used in the second sub-band region.
  • the channel decoding frequency domain decoding signal reconstructs the left and right channel frequency domain signals, since the code stream to be decoded only includes the mono coding layer and the stereo first enhancement layer code stream, and does not include the residual second subband region. Therefore, in the second sub-band region, the uncorrected decoding frequency domain decoding signal is used to reconstruct the left and right channel frequency domain signals, so that the decoding end and the encoding end signal are consistent, thereby improving the decoded stereo signal quality. . DRAWINGS
  • 1 is a flow chart of a parametric stereo audio encoding method
  • FIG. 2 is a flowchart of an audio decoding method in an embodiment of the present invention.
  • FIG. 3 is a flowchart of another audio decoding method in an embodiment of the present invention.
  • FIG. 4 is a schematic structural diagram of an audio decoder in an embodiment of the present invention.
  • FIG. 5 is a schematic structural diagram of an audio decoder according to an embodiment of the present invention.
  • the inventors of the present invention have found that the quality of the stereo signal reconstructed by the existing audio decoding method depends on two aspects: the reconstructed mono signal quality and the accuracy of the stereo parameter extraction. Among them, the mono signal quality reconstructed at the decoding end plays a very important role in the reconstructed stereo signal quality of the final output. Therefore, it is necessary to reconstruct the mono signal as high quality as possible on the decoding side. This basis can reconstruct high quality stereo signals.
  • the embodiment of the invention provides an audio decoding method, which can make the processing signals of the codec end consistent, so that the quality of the decoded stereo signal can be improved.
  • Embodiments of the present invention also provide corresponding audio decoders.
  • FIG. 1 a flowchart of the parametric stereo audio encoding method, specific steps as follows:
  • the frequency domain signals of the M signal and the S signal in the [0 ⁇ 7khz] frequency band are: ⁇ w(0), w(l), ---, w(N-l) ⁇ , ⁇ SXO ⁇ I N-1) ⁇ .
  • the frequency domain signals Z ⁇ /(0), /(l), ---, /(N-l) ⁇ in the [0 ⁇ 7khz] frequency band of the left and right channels are obtained.
  • the frequency domain signals of the left and right channels are divided into 8 subbands, and the left and right channel parameters ILD are extracted according to the subbands: W[band][l], W[band][r], and quantized and quantized.
  • Channel parameters ILD W q [band][l], W q [band][r], where bie (0,1,2,3,4,5,6,7), 1 indicates the left channel parameter ILD , r is identified as the right channel parameter ILD.
  • the frequency domain signal obtained by S13 is divided into eight sub-bands that are the same as the left and right channels, and the energy compensation parameters of the 5, 6, 7 sub-bands are calculated according to the formula (2), and the energy compensation parameters are quantized and encoded, and then quantized. .
  • Ecomp[band]
  • Unmofiyenergy[ban ] (/) xm x (/) respectively represent the original left channel i [st rt barui , end band ] in the current subband
  • t ED ⁇ ed(0), ed(l), '--, ed ⁇ N - 1) ⁇ perform hierarchical multi-quantization coding.
  • the coding information of the M signal is the most important, firstly packaged as a mono coding layer; channel parameters ILD, channel parameters ITD, energy adjustment factor, energy compensation parameters, KL transformation kernel and residual principal 0 ⁇ 4
  • the first quantization coded result is encapsulated as a stereo first enhancement layer; other information is also layered in importance.
  • the inventor of the present invention found in the research and practice of the prior art: In the case that only the mono coding layer and the stereo first enhancement layer code stream are received at the decoding end, the code stream to be decoded has only one tone.
  • the energy compensation for the decoding end is performed based on the energy-adjusted mono decoding frequency domain signal, and the encoding end step S14 extracts 5, 6, 7
  • the energy compensation parameters of the subband are based on the unresolved mono decoding frequency domain signal.
  • the processing signals of the codec segment are inconsistent, and the inconsistency of the codec signal causes the quality of the decoded output signal to appear. decline.
  • the decoding end determines the mono decoding frequency domain signal type used in the decoding process according to the state of the code stream to be decoded, when the decoding end only receives the mono coding layer and the stereo first enhancement layer code stream. Reconstructing the unresolved mono-decoded frequency-domain signal when reconstructing the stereo signals of the 5, 6, 7 sub-bands; using the energy-adjusted stereo signal when reconstructing the stereo signals of the 0 ⁇ 4 sub-bands The mono decoding frequency domain signal is reconstructed.
  • FIG. 2 it is a flowchart of an audio decoding method according to an embodiment of the present invention, including:
  • S21 Determine a code stream to be decoded as a mono coding layer and a stereo first enhancement layer code stream.
  • 522. Decode the mono coding layer to obtain a mono decoding frequency domain signal.
  • An embodiment of the present invention provides an audio decoding method, which determines a mono signal type used in reconstructing a monaural signal in a decoding process according to a received code stream state, and determines that the received code stream is In the mono coding layer and the stereo first enhancement layer code stream, the energy-adjusted mono decoding frequency domain signal is used in the first sub-band region to reconstruct the left and right channel frequency domain signals; in the second sub-band region The left and right channel frequency domain signals are reconstructed by using the unmodulated mono decoding frequency domain signal. Since the code stream to be decoded has only the mono coding layer and the stereo first enhancement layer code stream, the decoding end does not receive the signal stream.
  • the parameter of the second sub-band region of the residual so the left-channel frequency domain signal is reconstructed in the second sub-band region by using the un-encoded mono decoding frequency domain signal, so that the decoding end and the encoding end signal
  • the processed signals remain consistent, which improves the quality of the decoded stereo signal.
  • FIG. 3 is a flowchart of another audio decoding method according to an embodiment of the present invention, a specific step is described in detail below.
  • the decoding method adopted by the decoding end in the embodiment of the invention is described in detail below.
  • step S3 K determines whether the received code stream only contains the mono coding layer and the stereo first enhancement layer code stream, and if so, step S32;
  • the energy-adjusted mono decoding frequency domain signal M 2 ⁇ m 2 (0), w 2 (1), ..., w 2 (N - 1) ⁇ is obtained.
  • the first sub-band quantized residual information resleft qX ⁇ eleft ql (0) , eleft q (1), ⁇ , eleft ql ⁇ end ),0,0...,0 ⁇ , resright x ⁇ eright x (0), eright x (1), - - - , eright ⁇ end ),0,0 ⁇ . ⁇ ,0 ⁇ .
  • the energy-adjusted mono decoding frequency domain signal M 2 is used in the 0 ⁇ 4 sub-band, and the left and right channel frequency domain signals are reconstructed according to the equation (7), and the non-energy is used in the 5, 6, 7 sub-bands.
  • the adjusted mono decoding frequency domain signal ⁇ reconstructs the left and right channel frequency domain signals according to equation (8).
  • the energy adjustment is performed when reconstructing the stereo signals of the 0 ⁇ 4 subbands.
  • the mono decoding frequency domain signal M 2 reconstructs the left and right channel frequency domain signals.
  • the decoding end does not receive other enhancement layer code streams, so that the left and right channel residual information of the 5, 6, 7 sub-bands cannot be obtained, and
  • the energy compensation parameters of the 5, 6, 7 sub-bands are extracted according to the formula (2).
  • the energy compensation parameters are based on the mono decoding frequency domain signal ;; 3 ⁇ 4 lines, so in this step, when reconstructing the stereo signals of the 5, 6 and 7 sub-bands, the unresolved mono decoding frequency domain signal is used for reconstruction, and the stereo signals in the 0 ⁇ 4 sub-band are used.
  • the energy-modulated mono decoding frequency domain signal M 2 is reconstructed so that the signals at the codec end are consistent.
  • the frequency domain signal is divided into 8 subbands, and the 0 ⁇ 4 subbands of the principal element parameters are encapsulated in the stereo first enhancement layer, and other parameters related to the residual are encapsulated in other stereo enhancement layers for description. It should be noted that at this time, the 0 ⁇ 4 sub-band is called the first sub-band area, and the 5 ⁇ 7 sub-band is called the second sub-band area. It can be understood that, in a specific implementation, the parameter stereo sound The frequency domain signal can also be divided into other numbers of sub-bands during the frequency encoding process.
  • the embodiment of the present invention is at the decoding end at 0 ⁇ 3.
  • the subband reconstructs the left and right channel frequency domain signals using the energy-adjusted mono decoding frequency domain signal; the energy adjustment is performed in the 4-7 subband (second subband region)
  • the mono decoding frequency domain signal reconstructs the left and right channel frequency domain signals.
  • the mono signal type used in reconstructing the mono signal in the decoding process is determined according to the received code stream state, wherein the received code stream is determined to be mono.
  • the energy-adjusted mono decoding frequency domain signal is used to reconstruct the left and right channel frequency domain signals in the first sub-band region;
  • the energy-modulated mono decoding frequency domain signal reconstructs the left and right channel frequency domain signals. Since the code stream to be decoded has only the mono coding layer and the stereo first enhancement layer code stream, the decoder does not receive the residual error.
  • the parameters of the two sub-band regions so the left-channel frequency domain signal is reconstructed in the second sub-band region by using the energy-free mono decoding frequency domain signal, so that the processing signals of the decoding end and the encoding end signal are maintained. Consistent, which improves the quality of the decoded stereo signal.
  • the code stream received by the decoder includes other stereo enhancement layer code streams in addition to the mono coding layer and the stereo first enhancement layer code stream (for example, the mono coding layer and all stereo enhancement layer streams are completely received)
  • the decoding process is different from the above process. The difference is that the information of the residual in all sub-band regions can be decoded at this time, so in the frequency domain of the left and right channels The number (including the stereo signal of the first sub-band area and the stereo signal of the second sub-band area) is reconstructed using an energy-modulated mono decoding frequency domain signal. Moreover, since the information of the residual in all sub-band regions can be completely obtained, it is not necessary to perform energy compensation on the left and right channel frequency domain signals of the first sub-band or the second sub-band. Thereby the codec end processing signals are consistent.
  • the audio decoder 1 includes: a determining unit 41, a processing unit 42, and a first reconstructing unit 43.
  • the determining unit 41 is configured to determine whether the code stream to be decoded is a mono coding layer and a stereo first enhancement layer code stream, and if so, triggering the first reconstruction unit 43;
  • the processing unit 42 is configured to decode the mono coding layer to obtain a mono decoding frequency domain signal
  • the first reconstruction unit 43 is configured to reconstruct the left and right channel frequency domain signals by using the energy-adjusted mono decoding frequency domain signal in the first sub-band region; and adopting the processing unit 42 in the second sub-band region.
  • the decoded unchannelized frequency modulated frequency domain signal obtained by the decoding reconstructs the left and right channel frequency domain signals.
  • the processing unit 42 is further configured to: decode the stereo first enhancement layer code stream, obtain an energy adjustment factor, perform frequency peak analysis on the mono decoding frequency domain signal, and obtain a spectrum analysis result, according to the The spectrum analysis result and the energy adjustment factor perform energy adjustment on the mono decoded frequency domain signal.
  • a reconstruction unit 43 is specifically configured to use an energy-adjusted mono in the 0 ⁇ 4 sub-band
  • the channel decoding frequency domain signal reconstructs the left and right channel frequency domain signals, and the 5, 6, 7 subband uses the unenhanced mono channel decoding frequency domain signal decoded by the processing unit 42 to the left and right channel frequency domain signals. Refactoring.
  • the processing unit 42 is further configured to perform energy compensation on the 5, 6, 7 subbands of the reconstructed left and right channel frequency domain signals. Adjustment.
  • the energy-adjusted mono decoding frequency domain signal pair is used in the first sub-band region.
  • Reconstruction of the left and right channel frequency domain signals; reconstruction of the left and right channel frequency domain signals by the unadjusted mono frequency domain signal in the second subband region, since only the mono coding layer and the stereo are received The first enhancement layer code stream, so the parameters of the second sub-band region of the residual are not received, so the left-channel frequency domain signal is weighted by the un-enhanced mono-decoded frequency-domain signal in the second sub-band region. So that the decoding end and the encoding end process the signal to be consistent, so the quality of the decoded stereo signal can be improved.
  • FIG. 4 is a schematic structural diagram of an audio decoder according to an embodiment of the present invention, which is different from the audio decoder 1 in that the audio decoder 2 further includes a second reconstruction unit 51, where:
  • the second reconstruction unit 51 uses The left and right channel frequency domain signals are reconstructed by using the energy-adjusted mono-decoded frequency domain signal in all sub-band regions.
  • first reconstruction unit 43 and the second reconstruction unit 51 can be integrated as one reconstruction unit.
  • the storage medium may include: a ROM, a RAM, a magnetic disk or an optical disk, and the like.

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Description

一种音频解码方法和音频解码器 本申请要求于 2009 年 5 月 14 日提交中国专利局、 申请号为 200910137565.3,发明名称为 "一种音频解码方法和音频解码器" 的中国专 利申请的优先权, 在先申请文件的内容通过引用结合在本申请中。 技术领域
本发明涉及多声道音频编解码技术领域, 具体涉及一种音频解码方法 和音频解码器。 背景技术
目前, 多声道音频信号有着广泛的应用场景, 如电话会议、 游戏等, 因此多声道音频信号的编解码也越来越受到重视。 基于波形编码的传统编 码器如 MPEG-II ( Moving Pictures Experts Group II, 动态图像专家组标准第 二版)、 MP3 ( Moving Picture Experts Group Audio Layer III, 动态图像专家 组音频第三层)和 AAC ( Advanced Audio Coding, 高级音频编码)在对多 声道信号进行编码时, 都是对每一个声道进行独立编码。 这种方法虽然能 够较好地恢复出多声道信号, 但是需要的带宽、 编码码率是单声道信号的 数倍。
目前较为流行的立体声或多声道编码技术是参数立体声编码, 其利用 很少的带宽就可以重建出听觉感受和原始信号完全相同的多声道信号。 其 基本方法是: 在编码端, 将多声道信号下混成一个单声道信号, 并对此信 号进行独立编码, 同时提取各声道间的声道参数, 并对这些参数进行编码。 在解码端, 首先解码出下混后的单声道信号, 然后解码出各声道间的声道 参数, 最后利用这些声道参数与下混后的单声道信号一起合成出各多声道 信号。 典型的参数立体声编码技术, 如 PS (变量立体声)等都有着广泛的 应用。 在参数立体声编码中通常用来描述各声道间相互关系的声道参数有
ITD( Inter-channel Time Difference,声道间时间差)、 ILD( Inter-channel Level Difference, 声道间幅度差)及 ICC ( Inter-Channel Coherence, 声道间相关 性)等。 这些参数可以表征立体声声像信息, 如声源发声方向、 位置等。 在编码端对这些参数进行编码传输, 并且对由多声道得到的下混信号进行 编码传输, 就可以在解码端较好地重构出立体声信号, 而且占用带宽小, 编码码率低。 但是, 在对现有技术的研究和实践过程中, 本发明的发明人发现, 采 用现有的参数立体声编解码方法, 存在编解码端处理信号不一致的问题, 这种编解码信号的不一致会使解码得到的信号质量下降。
发明内容
本发明实施例提供一种音频解码方法和音频解码器, 能够使编解码端 处理信号一致, 提高解码立体声信号的质量。
本发明实施例包括以下技术方案:
一种音频解码方法, 包括:
确定待解码的码流为单声道编码层和立体声第一增强层码流; 对所述单声道编码层进行解码, 获得单声道解码频域信号;
在第一子带区域采用能量调整后的所述单声道解码频域信号对左右声 道频域信号进行重构;
在第二子带区域采用未经能量调整的所述单声道解码频域信号对左右 声道频域信号进行重构。
一种音频解码器, 包括: 判断单元、 处理单元和第一重构单元, 其中: 所述判断单元, 用于判断待解码的码流是否为单声道编码层和立体声 第一增强层码流, 如果是, 则触发第一重构单元;
所述处理单元, 用于对所述单声道编码层进行解码, 获得单声道解码 频域信号;
所述第一重构单元, 用于在第一子带区域采用能量调整后的单声道解 码频域信号对左右声道频域信号进行重构; 在第二子带区域采用所述处理 单元解码得到的未经能量调整的所述单声道解码频域信号对左右声道频域 信号进行重构。 本发明实施例根据待解码的码流状态决定解码过程中在对单声道信号 进行重构时所采用的单声道信号类型, 其中在确定待解码的码流为单声道 编码层和立体声第一增强层码流时, 在第一子带区域采用能量调整后的单 声道解码频域信号对左右声道频域信号进行重构; 在第二子带区域采用未 经能量调整的单声道解码频域解码信号对左右声道频域信号进行重构, 由 于待解码的码流只包含单声道编码层和立体声第一增强层码流, 而不包含 残差第二子带区域的参数, 所以在第二子带区域采用未经能量调整的解码 频域解码信号对左右声道频域信号进行重构, 从而使得解码端与编码端信 号保持一致, 因此可以提高解码立体声信号质量。 附图说明
图 1是参数立体声音频编码方法流程图;
图 2是本发明实施例中一种音频解码方法流程图;
图 3是本发明实施例中另一种音频解码方法流程图;
图 4是本发明实施例中音频解码器一结构示意图;
图 5是本发明实施例中音频解码器二结构示意图。
具体实施方式
本发明的发明人发现, 现有音频解码方法所重构的立体声信号质量取 决于两方面: 重构的单声道信号质量和立体声参数提取的准确性。 其中, 在解码端重构的单声道信号质量对最终输出的重构立体声信号质量起着非 常重要的作用。 因此在解码端需要尽可能高质量地重构出单声道信号, 在 此基 上才能重构出高质量的立体声信号。
本发明实施例提供一种音频解码方法, 能够使编解码端的处理信号一 致, 从而可以提高解码立体声信号的质量。 本发明实施例还提供相应的音 频解码器。
为使本领域技术人员更好地理解和实现本发明实施例, 以下首先对参 数立体声编码在编码端所执行的操作进行伴细说明, 参照图 1, 为参数立体 声音频编码方法流程图, 具体步骤如下:
511、根据原始左右声道信号提取声道参数 ITD,根据 ITD参数对左右声 道信号进行声道延时调整, 对调整后的左右声道信号进行下混处理, 得到 单声道信号(也可称为和信号即 M信号)和边信号(S信号)。
M信号和 S信号在 [0~7khz]频带 内 的频域信号分别 为 : {w(0),w(l),---,w(N-l)}, ^SXO^I N— 1)}。 根据式(1 )得到左右 声 道 在 [0~7khz] 频 带 内 的 频 域 信 号 Z{/(0),/(l),---,/(N-l)} ,
R{r(0),r(\),---,r(N-\)}a
Figure imgf000006_0001
512、 将左右声道的频域信号划分为 8个子带, 按子带提取左右声道参 数 ILD: W[band][l],W[band][r] , 并进行量化编码得到量化后的声道参数 ILD: Wq[band][l],Wq[band][r], 其中 b i e (0,1,2,3,4,5,6,7), 1表示左声道参 数 ILD, r标识为右声道参数 ILD。
513、 对 M信号进行编码, 并且进行本地解码得到本地解码频域信号 Mx {mx (0), mx (1),•••,w1(N-l)}0
514、将 S13得到的 频域信号划分为和左右声道相同的 8个子带,按照 式(2)计算 5, 6, 7子带的能量补偿参数 画 对能量补偿参数进 行量化编码, 得到量化后
Figure imgf000006_0002
。 ecomp[band] =
Figure imgf000007_0001
q an r x q an r x nmo yenergy an
(2) 其中: C[ba"i] [/][/]= ,
Figure imgf000007_0002
Unmofiyenergy[ban ] = (/) x mx (/)分别表示在当前子带原始左声道 i [st rtbarui,endband]
能量、 原始右声道能量、 本地解码单声道能量, [stoW , i ]表示当前 子带频率点的起始位置和结束位置。
515、 对本地解码频域信号 进行频语峰值分析, 得到频谱分析结果 MASK{mask{Q),mask{\),- --,mask{N - 1)}, 其中 mask(i)G {0,1}。 当 ]^在1处的 频语信号1¾为峰值时, wo^( ) = l, 否则 wo^0') = 0。
516、 选择最佳能量调整因子 multiplier, 按照式(3)对解码频域信号 Mi 进 行 能 量 调 整 , 得 到 能 量 调 整 后 的 频 域 信 号 M2 {m2(0),m2(\),- · ·, w2(N - 1)}, 对能量调整因子 multiplier进行量化编码。
Figure imgf000007_0003
S17、 利用能量调整后的频域信号 M2、 左右声道频域信号 L、 R以及左 右声道量化后的声道参数 ILD: Wq, 按照式 (4)计算左右声道残差信息
Figure imgf000007_0004
--,eleft(N - 1) , 以 及 resright{eright(0), eright(\), ···, eright(N - 1)}。
eleft(i) = /(/) - W [band] [I] x m2 (i)
, i ^,end , band = 0,1,2,3,·· -7 erightii) = r{i)-Wq [band] [r]xm2 (i) 1 band, band ,, , , (4)
S18、 对左右声道残差进行 K-L (Karhunen-Loeve)变换, 对变换核 H进 行量化编码, 对变换后得到的残差主元^7{^(0),^(1),...,^(^ - 1)}、 残差
、 t ED{ed(0),ed(l),' - -,ed{N - 1)}进行分层多次量化编码。
S 19、 对编码端提取的各种编码信息按照重要程度进行分层封装码流, 将编码码流传输。
其中, M信号的编码信息最重要, 首先作为单声道编码层进行封装; 声 道参数 ILD、 声道参数 ITD、 能量调整因子、 能量补偿参数、 K-L变换核和 残差主元 0~4子带第一次量化编码结果作为立体声第一增强层进行封装; 其 他信息也按重要性进行分层封装。
由于码流的传输网络环境时刻在变化, 当网络资源不足时, 在解码端 不能接收到所有的编码信息。 例如只接收到单声道编码层和立体声第一增 强层码流, 其他层码流没有接收到。
本发明的发明人在对现有技术的研究和实践过程中发现: 对于解码端 只接收到单声道编码层和立体声第一增强层码流的情况下, 即待解码的码 流只有单声道编码层和立体声第一增强层码流, 现有技术中对解码端的能 量补偿是基于能量调整后的单声道解码频域信号进行的, 而在编码端步骤 S14中提取 5, 6, 7子带的能量补偿参数是基于未经能量调整的单声道解码 频域信号进行的, 此时, 编解码段的处理信号不一致, 这种编解码端信号 的不一致会使解码输出信号的质量出现下降。
而本发明实施例在解码端根据待解码的码流状态决定解码过程中采用 的单声道解码频域信号类型, 当解码端只接收到单声道编码层和立体声第 一增强层码流时, 在重构 5, 6, 7子带的立体声信号时采用未经能量调整的 单声道解码频域信号进行重构; 在重构 0~4子带的立体声信号时采用经过能 量调整后的单声道解码频域信号进行重构。
参照图 2, 为本发明实施例中一种音频解码方法流程图, 包括:
S21、 确定待解码的码流为单声道编码层和立体声第一增强层码流; 522、 对所述单声道编码层进行解码, 获得单声道解码频域信号;
523、在第一子带区域采用能量调整后的所述单声道解码频域信号对左 右声道频域信号进行重构;
524、在第二子带区域采用未经能量调整的所述单声道解码频域信号对 左右声道频域信号进行重构。
本发明实施例提供了一种音频解码方法, 根据接收到的码流状态决定 解码过程中在对单声道信号进行重构时所采用的单声道信号类型, 在确定 接收到的码流为单声道编码层和立体声第一增强层码流时, 在第一子带区 域采用能量调整后的单声道解码频域信号对左右声道频域信号进行重构; 在第二子带区域采用未经能量调整的单声道解码频域信号对左右声道频域 信号进行重构, 由于待解码的码流只有单声道编码层和立体声第一增强层 码流, 解码端没有接收到残差第二子带区域的参数, 所以在第二子带区域 采用未经能量调整的单声道解码频域信号对左右声道频域信号进行重构, 从而使得解码端与编码端信号的处理信号保持一致, 从而可以提高解码立 体声信号质量。
参照图 3, 为本发明实施例中另一种音频解码方法流程图, 以下通过具 体步骤详细说明在解码端确定只接收到单声道编码层和立体声第一增强层 码流的情况下, 本发明实施例在解码端所采用的解码方法:
S3 K判断接收到的码流是否只包含单声道编码层和立体声第一增强层 码流, 如果是, 则执行步骤 S32;
S32、对接收到的单声道编码层码流可以采用与编码端使用的音频 /语音 编码器对应的任意一种音频 /语音解码器进行解码操作, 进行得到单声道解 码频域信号:
Figure imgf000009_0001
该信号即为编码端步骤 S13得到 的信号。 从立体声第一增强层码流中读取各个参数对应的码字, 对各参数 进行解码得到声道参数 ILD: W band][l],W band][r] . 声道参数 ITD、 能量 调整因子 multiplie 量化后能量补偿参数 ecowpjb i]、 K-L变换核 H和残 差主元 0~4子带第一次量化结果 EUq {euqX (0), euqX (1), ···, euqX {end, ),0,0…,0}。
533、 对单声道解码频域信号 Ml进行频语峰值分析, 即在频域中搜索 频语极大值, 得到频谱分析结果: MASK maskiQ maski^cmask N— 1 , 其中 wa^()e{0,l}。 当 Ml在 i处的频谱信号 ml(i)为峰值, 即极大值时, mask{i) = 1, 否贝 "] mask{i) = 0。
534、 根据解码得到的能量调整因子 multiplier和频谱分析结果对单声道 解码频域信号采用式(5)进行能量调整:
(i) x multiplier, mask i) = 0
Figure imgf000010_0001
ml (i) , mask{i) = 1 (5)
从 而 得 到 能 量 调 整 后 的 单 声 道 解 码 频 域 信 号 M2 {m2 (0), w2 (1),…, w2 (N - 1)}。
S35、 根据 K-L变换核 H和残差主元 0~4子带第一次量化结果 {e"l(0),eM l(l), 0"c/4 ),0,0…,。)按式( 6 )进行^ K-L变换,得到左右声 道 在 0~4 子 带 的 第 一 次 量 化 残 差 信 息 resleft qX {eleftql (0), eleftq (1), ···, eleftql {end ),0,0…,0} , resright x {eright x (0), eright x (1), - - - , eright χ end ),0,0· . ·,0}。
(6)
Figure imgf000010_0002
S36、 在 0~4子带采用经过能量调整后的单声道解码频域信号 M2, 根据 式(7) 重构左右声道频域信号, 在 5, 6, 7子带采用未经能量调整的单声 道解码频域信号^^根据式(8)重构左右声道频域信号。
I ( = eleftql ( + Wq [band] [I] x m2 (i)
r i) = eright , (i) + W [band] [r] x m2 (i) i [st rt band, end band],band = 0,1,2,3,4
( 7 )
/'(/') = deft (i) + W [band] [I] x mx (i)
, i≡\starth , , end, A.band = 5,6,7 r ' ( = erightql (/) + Wq [band] [r] x mi (/) L , 」, ,,
( 8 ) 由于在解码端接收到了立体声第一增强层码流, 其中包含 0-4子带的左 右声道残差信息, 因此在重构 0~4子带的立体声信号时采用能量调整后的单 声道解码频域信号 M2对左右声道频域信号进行重构。 而除了单声道编码层 和立体声第一增强层之外的码流, 解码端没有接收到其他的增强层码流, 从而无法获得 5, 6, 7子带的左右声道残差信息, 且在编码端的步骤 S14中, 是按照式(2 )提取 5, 6, 7子带的能量补偿参数的, 从 S14可以看出, 所述 能量补偿参数是基于单声道解码频域信号 Μι;¾行的, 因此本步骤中在重构 5, 6 , 7子带的立体声信号时采用未经能量调整的单声道解码频域信号 进行重构, 而在 0~4子带的立体声信号采用经过能量调整后的单声道解码频 域信号 M2进行重构, 从而使得编解码端的信号保持一致。
537、 按照式(9 )对重构后的左右声道频域信号的 5, 6, 7子带进行能 量补偿调整。
_ χ J Qecompq [band]/20
, , .、― ,, .、 1 A_[ ]/20 - 1 G startband , endband ] , band = 5,6,7
(9)
538、 对左右声道频域信号进行处理, 得到最终的左右声道输出信号。 以上以参数立体声音频编码过程中将频域信号划分为 8个子带, 且主元 参数的 0~4子带封装在立体声第一增强层, 有关残差的其他参数封装在其他 立体声增强层进行说明, 需要说明的是,此时, 0~4子带称为第一子带区域, 5~7子带称为第二子带区域。 可以理解的是, 在具体实施中, 参数立体声音 频编码过程中也可以将频域信号划分为其他数目的多个子带。 即使对于划 将主元参数的 0~3子带封装在立体声第一增强层, 有关残差的其他参数封装 在其他立体声增强层, 此时, 0~3子带称为第一子带区域, 4~7子带称为第 二子带区域, 相应的, 对于待解码的码流只有单声道编码层和立体声第一 增强层码流的情况, 本发明实施例在解码端在 0~3子带(第一子带区域)采 用能量调整后的单声道解码频域信号对左右声道频域信号进行重构; 在 4~7 子带 (第二子带区域)采用未经能量调整的单声道解码频域信号对左右声 道频域信号进行重构。
从本实施例可以看出, 根据接收到的码流状态决定解码过程中在对单 声道信号进行重构时所采用的单声道信号类型, 其中在确定接收到的码流 为单声道编码层和立体声第一增强层码流时, 在第一子带区域采用能量调 整后的单声道解码频域信号对左右声道频域信号进行重构; 在第二子带区 域采用未经能量调整的单声道解码频域信号对左右声道频域信号进行重 构, 由于待解码的码流只有单声道编码层和立体声第一增强层码流, 解码 端没有接收到残差第二子带区域的参数, 所以在第二子带区域采用未经能 量调整的单声道解码频域信号对左右声道频域信号进行重构, 从而使得解 码端与编码端信号的处理信号保持一致, 从而可以提高解码立体声信号质 量。
对于解码端接收到的码流除了单声道编码层和立体声第一增强层码流 外, 还包含其他立体声增强层码流(例如, 单声道编码层和所有立体声增 强层码流完全接收) 时, 解码过程与上述过程有所不同。 不同之处在于, 此时可以解码得到残差在所有子带区域的信息, 因此在对左右声道频域信 号 (包括第一子带区域的立体声信号和第二子带区域的立体声信号)进行 重构时采用能量调整后的单声道解码频域信号。 并且, 由于能够完整得到 残差在所有子带区域的信息, 因此不需要对第一子带或第二子带的左右声 道频域信号进行能量补偿。 从而使得编解码端处理信号一致。
以上对本发明实施例所采用的音频解码方法进行了详细说明, 以下对 使用上述音频解码方法的解码器进行对应介绍。
参照图 4, 为本发明实施例中音频解码器一结构示意图, 音频解码器一 包括: 判断单元 41、 处理单元 42和第一重构单元 43, 其中:
判断单元 41, 用于判断待解码的码流是否为单声道编码层和立体声第 一增强层码流, 如果是, 则触发第一重构单元 43;
处理单元 42, 用于对所述单声道编码层进行解码, 获得单声道解码频 域信号;
第一重构单元 43, 用于在第一子带区域采用能量调整后的单声道解码 频域信号对左右声道频域信号进行重构; 在第二子带区域采用所述处理单 元 42解码得到的未经能量调整的所述单声道解码频域信号对左右声道频域 信号进行重构。
所述处理单元 42还用于对所述立体声第一增强层码流进行解码, 获得 能量调整因子, 对所述单声道解码频域信号进行频语峰值分析, 获得频谱 分析结果, 根据所述频谱分析结果和所述能量调整因子对所述单声道解码 频域信号进行能量调整。
如果参数立体声音频编码过程中将频域信号划分为 8个子带, 且主元参 数的 0~4子带封装在立体声第一增强层, 有关残差的其他参数封装在其他立 体声增强层, 则第一重构单元 43具体用于在 0~4子带采用能量调整后的单声 道解码频域信号对左右声道频域信号进行重构, 在 5, 6, 7子带采用处理单 元 42解码得到的未经能量调整的单声道解码频域信号对左右声道频域信号 进行重构。
当第一重构单元 43获得重构后的左右声道频域信号后, 所述处理单元 42还用于对重构后的左右声道频域信号的 5, 6, 7子带进行能量补偿调整。
可见, 本实施例所介绍的音频解码器在确定只接收到单声道编码层和 立体声第一增强层码流时, 在第一子带区域采用能量调整后的单声道解码 频域信号对左右声道频域信号进行重构; 在第二子带区域采用未经能量调 整的单声道频域信号对左右声道频域信号进行重构, 由于只接收到单声道 编码层和立体声第一增强层码流, 因此残差第二子带区域的参数没有接收 到, 所以在第二子带区域采用未经能量调整的单声道解码频域信号对左右 声道频域信号进行重构, 从而使得解码端与编码端处理信号保持一致, 因 此可以提高解码立体声信号质量。
参照图 4, 为本发明实施例中音频解码器二结构示意图, 与音频解码器 一的不同之处在于, 音频解码器二中还包括第二重构单元 51, 其中:
当所述判断单元 41的判断结果为待解码的码流除了单声道编码层和立 体声第一增强层码流外, 还包含其他立体声增强层码流时, 所述第二重构 单元 51用于在所有子带区域采用能量调整后的所述单声道解码频域信号对 左右声道频域信号进行重构。
可以理解的是, 在具体实施中, 第一重构单元 43与第二重构单元 51可 以集成在一起, 作为一个重构单元。
本领域普通技术人员可以理解上述实施例的各种方法中的全部或部分 步骤是可以通过程序来指令相关的硬件来完成, 该程序可以存储于一计算 机可读存储介质中, 存储介质可以包括: ROM、 RAM, 磁盘或光盘等。 以上对本发明实施例所提供的音频解码方法和音频解码器进行了详细 上实施例的说明只是用于帮助理解本发明的方法及其核心思想; 同时, 对 于本领域的一般技术人员, 依据本发明的思想, 在具体实施方式及应用范 围上均会有改变之处, 综上所述, 本说明书内容不应理解为对本发明的限 制。

Claims

权利要求
1、 一种音频解码方法, 其特征在于, 包括:
确定待解码的码流为单声道编码层和立体声第一增强层码流; 对所述单声道编码层进行解码, 获得单声道解码频域信号;
在第一子带区域采用能量调整后的所述单声道解码频域信号对左右声 道频域信号进行重构;
在第二子带区域采用未经能量调整的所述单声道解码频域信号对左右 声道频域信号进行重构。
2、 如权利要求 1所述的方法, 其特征在于, 还包括:
对所述单声道解码频域信号进行能量调整。
3、 如权利要求 2所述的方法, 其特征在于, 所述对所述单声道解码频 域信号进行能量调整包括:
对所述立体声第一增强层码流进行解码, 获得能量调整因子; 对所述单声道解码频域信号进行频语峰值分析, 获得频谱分析结果; 根据所述频谱分析结果和所述能量调整因子对所述单声道解码频域信 号进行能量调整。
4、如权利要求 1-3任一所述的方法, 其特征在于, 所述在第一子带区域 采用能量调整后的所述单声道解码频域信号对左右声道频域信号进行重 构; 在第二子带区域采用未经能量调整的所述单声道解码频域信号对左右 声道频域信号进行重构具体为:
在 0~4子带采用能量调整后的所述单声道解码频域信号对左右声道频 域信号进行重构; 在 5, 6, 7子带采用未经能量调整的所述单声道解码频域 信号对左右声道频域信号进行重构。
5、 如权利要求 4所述的方法, 其特征在于, 对左右声道频域信号重构 后还包括: 对重构后的左右声道频域信号的 5, 6, 7子带进行能量补偿调整。
6、 一种音频解码器, 其特征在于, 包括: 判断单元、 处理单元和第一 重构单元, 其中:
所述判断单元, 用于判断待解码的码流是否为单声道编码层和立体声 第一增强层码流, 如果是, 则触发第一重构单元;
所述处理单元, 用于对所述单声道编码层进行解码, 获得单声道解码 频域信号;
所述第一重构单元, 用于在第一子带区域采用能量调整后的单声道解 码频域信号对左右声道频域信号进行重构; 在第二子带区域采用所述处理 单元解码得到的未经能量调整的所述单声道解码频域信号对左右声道频域 信号进行重构。
7、 如权利要求 6所述的音频解码器, 其特征在于, 所述处理单元还用 于对所述立体声第一增强层码流进行解码, 获得能量调整因子, 对所述单 声道解码频域信号进行频谱峰值分析, 获得频谱分析结果, 根据所述频谱 分析结果和所述能量调整因子对所述单声道解码频域信号进行能量调整。
8、 如权利要求 7所述的音频解码器, 其特征在于, 所述第一重构单元 具体用于在 0~4子带采用能量调整后的单声道解码频域信号对左右声道频 域信号进行重构; 在 5, 6, 7子带采用所述处理单元解码得到的未经能量调 整的所述单声道解码频域信号对左右声道频域信号进行重构。
9、 如权利要求 8所述的音频解码器, 其特征在于, 当第一重构单元获 得重构后的左右声道频域信号后, 所述处理单元还用于对重构后的左右声 道频域信号的 5, 6, 7子带进行能量补偿调整。
10、 如权利要求 6所述的音频解码器, 其特征在于, 还包括: 第二重构 单元, 当所述判断单元的判断结果为待解码的码流除了单声道编码层和立体 声第一增强层码流外, 还包含其他立体声增强层码流时, 所述第二重构单 元用于在所有子带区域采用能量调整后的所述单声道解码频域信号对左右 声道频域信号进行重构。
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