WO2008089705A1 - Encoding and decoding method andapparatus - Google Patents

Encoding and decoding method andapparatus Download PDF

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Publication number
WO2008089705A1
WO2008089705A1 PCT/CN2008/070170 CN2008070170W WO2008089705A1 WO 2008089705 A1 WO2008089705 A1 WO 2008089705A1 CN 2008070170 W CN2008070170 W CN 2008070170W WO 2008089705 A1 WO2008089705 A1 WO 2008089705A1
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Prior art keywords
data
frame
decoding
length
coding
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.)
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PCT/CN2008/070170
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English (en)
French (fr)
Inventor
Wei Guo
Peilin Liu
Wei Li
Lijing Xu
Qing Zhang
Jianfeng Xu
Shenghu Sang
Zhengzhong Du
Chen Hu
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Huawei Technologies Co Ltd
Shanghai Jiao Tong University
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Huawei Technologies Co Ltd
Shanghai Jiao Tong University
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Priority to JP2009546636A priority Critical patent/JP2010517083A/ja
Priority to EP08706547A priority patent/EP2120233A4/en
Publication of WO2008089705A1 publication Critical patent/WO2008089705A1/zh
Priority to US12/508,103 priority patent/US20090299757A1/en
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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Classifications

    • 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
    • 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
    • G10L19/022Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring
    • 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
    • G10L19/0212Speech 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
    • 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

Definitions

  • the present invention relates to the field of codec technology, and in particular, to a codec method and apparatus.
  • the AMR-WB+ based encoder mainly includes the following two encoding modes:
  • Excitation transform code excitation mode, used to encode tones.
  • the AMR-WB+ technology is formed by extending the low-rate speech coding, specifically a hybrid coding method combining ACELP coding for speech and TCX coding for tone.
  • the currently selected coding mode is determined by comparing the SEGSNR (Segmented Signal to Noise Ratio) values of the two modes.
  • SEGSNR Segmented Signal to Noise Ratio
  • AMR-WB+ converts from ACELP encoded frame to TCX encoded frame. An implementation that removes the zero input response is used.
  • the ACELP coding mode is used, and the current frame is switched by the TCX coding mode.
  • the corresponding TCX coding implementation process is shown in Figure 1.
  • the input audio signal is first passed.
  • the ZIR (zero input response) after windowing is subtracted from the perceptually weighted signal, and then an adaptive window is added to the obtained signal, and transform domain coding is performed to Obtain the corresponding TCX coded stream.
  • An embodiment of the present invention provides a codec method and apparatus, so that the complexity of the process of eliminating inter-frame discontinuity is reduced in the codec process, thereby reducing the implementation complexity of the codec process. .
  • An embodiment of the present invention provides an encoding method, including:
  • An embodiment of the present invention provides an encoding apparatus including an encoding mode switching identification unit, a previous encoding frame overlap data acquiring unit, and a second encoding unit, where:
  • the coding mode switching identification unit performs coding in the first coding mode after determining that the previous frame is used, and the current frame needs to be encoded in the transform domain coding mode, and triggers the acquisition of the previous coded frame overlapping data. unit; [19]
  • the previous coded frame overlap data acquisition unit is configured to obtain, according to the determined data length of the current frame and the coded data of the previous frame, the corresponding length of the coded data in the previous frame, and provide the coded data Two coding unit;
  • the second coding unit performs overlapping processing on the coded data acquired by the previous coded frame overlap data acquiring unit and the coded data of the current frame to obtain a coded result.
  • An embodiment of the present invention provides a decoding method, including:
  • the decoded data obtained from the previous frame is overlapped with the decoded data of the current frame to obtain a decoding result.
  • An embodiment of the present invention provides a decoding apparatus including a decoding mode switching identification unit, a previous decoding frame overlapping data acquiring unit, and a second decoding unit, where:
  • a decoding mode switching identifying unit configured to determine, according to information in the decoded code stream, that the previous frame is decoded by using the first decoding mode, and the current frame is obtained by decoding the second code decoding mode, before triggering a decoding frame overlap data acquiring unit;
  • the previous decoded frame overlap data acquiring unit is configured to obtain, according to the determined data length of the decoded data of the current frame and the previous frame, the decoded data of the corresponding length in the corresponding previous frame, and provide the data to the first Two decoding unit;
  • the second decoding unit is configured to overlap the decoded data acquired by the previous decoded frame overlap data acquiring unit and the decoded data of the current frame to obtain a decoding result.
  • the embodiments provided by the present invention can implement corresponding mode switching codec processing without using a filter operation, so that the operation complexity of the entire codec process is not High, and thus easier to implement hardware and software.
  • the embodiment provided by the present invention can also effectively ensure the effect of eliminating discontinuity between frames in the case where the zero input response is not similar to the signal at the beginning of the current frame.
  • FIG. 1 is a schematic block diagram of a TCX encoding process in the prior art
  • FIG. 2 is a schematic block diagram of a TCX decoding process in the prior art
  • FIG. 3 is a domain window function applied in the process of calculating the ZIR value after windowing processing in the prior art.
  • FIG. 4 is a schematic block diagram of a TCX encoding process in an embodiment of the present invention.
  • FIG. 5 is a schematic block diagram of a TCX decoding process according to an embodiment of the present invention.
  • FIG. 6 is a schematic structural diagram of an input voice frame according to an embodiment of the present invention.
  • FIG. 7 is a schematic view showing a window shape after windowing processing according to an embodiment of the present invention.
  • FIG. 8 is a schematic diagram of frame-to-frame overlap smoothing in a decoding process according to an embodiment of the present invention.
  • FIG. 9 is a schematic structural diagram of a specific implementation of a codec apparatus according to an embodiment of the present invention.
  • the embodiment of the encoding provided by the present invention includes: encoding the first encoding mode in determining the previous frame, and further determining that the current frame needs to be encoded by using the second encoding mode, that is, determining that the encoding process occurs. And switching the coding mode of the coding mode according to the current frame and the coded data of the previous frame and the subsequent frame, and obtaining the coded data of the overlapped data length in the corresponding previous frame and the subsequent frame. And performing encoding processing based on the second encoding mode on the encoded data obtained from the data of the previous frame and the subsequent frame and the encoded data of the current frame to obtain a coding result.
  • the data length of the overlap portion is determined according to the frame length of each coded frame and is preset in the encoder, and the longer the frame length of the coded frame is, the longer the data length of the corresponding overlap portion is. .
  • the data length of the overlapped portion of the encoded data of the current frame and the previous frame is the first length
  • the encoded data of the current frame and the subsequent frame overlap is the second length.
  • the first length may be the same as the second length.
  • the corresponding embodiment is not limited to the two length values being the same in a specific application process.
  • the first coding mode may be specifically, but not limited to, a linear predictive coding mode
  • the second coding mode may be, but not limited to, a transform domain coding; further, a corresponding implementation
  • the example can be applied to the mode switching encoding process between various linear predictive coding and transform domain coding, for example, mode switching between ACELP coding and TCX coding, and the like.
  • the decoding provided by the present invention includes: performing decoding operation on the received code stream, and determining The previous frame in the received code stream is obtained by decoding in the first decoding mode, and after the current frame is decoded in the second decoding mode, the overlap between the determined current frame and the decoded data of the previous frame is determined. Part of the data length, the decoded data of the overlapped data length in the previous frame is obtained; and the decoded data obtained from the previous frame is overlapped with the decoded data of the current frame, specifically The decoded data of the overlapping portion data length in the frame is subjected to windowing superposition processing with the header of the decoded data of the current frame to obtain a decoding result.
  • an embodiment of the present invention proposes an overlap smoothing technique of ACELP and T CX coding mode switching, which can obtain better performance while ensuring a constant code rate. Inter-frame smoothing effect. Moreover, in the process of applying the embodiment, it is not necessary to perform complex synthetic perceptual weighting filter calculation, so that the corresponding computational complexity is reduced compared with the inter-mode smoothing technique of AMR-WB+ in the prior art.
  • the embodiment of the present invention mainly uses the inter-mode overlap smoothing technology to implement the effect of switching between two encoding modes, and the encoding is performed by the TCX codec processing to improve the TCX. Codec efficiency, reducing the complexity of TCX codec.
  • the TCX codec technical solution based on the inter-mode overlapping smoothing technique provided by the present invention will be described below.
  • the synthesized audio signal is obtained through the inverse perceptual weighting filter; wherein, if the previous frame uses the ACELP encoding mode, then at the decoding end, Using the processing strategy corresponding to the encoding end, the overlapping result of the decoding result of the previous frame is overlapped with the current frame to obtain the decoding result of the current frame; referring to the example in the encoding process, the TCX decoder is used.
  • the window overlap is overlapped with the last overlap of the current frame and the last subframe of the previous frame of the ACELP composite signal to obtain a final synthesized audio signal.
  • the specific inter-frame overlap techniques include:
  • the last several audio data processed by the previous frame ACELP (for example, may be 16, 32 or 64 points) s peech data) TCX encoding together with current frame audio data, the last several pieces of audio data refer to audio data determined according to the length of the encoding frame and overlapping the data length of the previous frame;
  • L_frame represents the frame length of the current frame TCX encoding, which may be 256, 512 or 1024 respectively corresponding to the three encoding modes of TCX.
  • L1 represents the length of the audio signal that is overlapped with the previous frame
  • L2 is the number of points of the audio signal taken for ove rlap (overlap) with the next frame
  • L represents the actual processing of the current frame.
  • the length of the audio signal; the value of each parameter in Figure 6 can be:
  • the length of the current frame overlapping with the previous frame varies with the change of the TCX coding mode, and has an adaptive effect.
  • Peer the actual frame length and A of the speech (voice) signal per frame of the TCX processed by this method
  • the actual frame length matching in MR-WB+ ensures the correctness of the encoding.
  • w(n) refers to the curve of the portion shown by L2 in Fig. 7; that is, the portion overlapping the previous frame is not windowed, and the cosine is added to the portion overlapping with the latter frame.
  • Window w(n) refers to the curve of the portion shown by L2 in Fig. 7; that is, the portion overlapping the previous frame is not windowed, and the cosine is added to the portion overlapping with the latter frame.
  • the window length of the cosine window is only AMR-WB
  • the window length added by the frame header of the next frame should be the same as the length of L2, that is, the length of the corresponding overlap part should be consistent with the current frame to ensure the smoothing effect between frames.
  • the TCX decoder decodes the synthesized audio signal of the current frame from the received current frame TCX encoded code stream, and overlaps the header with the decoded audio signal of the previous frame ACELP. Windowing is done to superimpose, resulting in the final synthesized audio output.
  • a triangular window is used in the overlapping portion, and the synthesized audio signal of the last L1 samples of the ACELP is represented by w2(n), and the synthesized audio signal of the overlapping portion of the TCX is used by wl. (n) indicates that the corresponding synthesized audio signal is as follows:
  • the present invention further provides an embodiment of a codec device, which is specifically shown in FIG. 9, and specifically includes an encoding device and a decoding device.
  • a codec device which is specifically shown in FIG. 9, and specifically includes an encoding device and a decoding device.
  • the specific implementation structures of the two devices will be described below.
  • the device specifically includes an encoding mode switching identification unit, a previous encoding frame overlapping data acquiring unit, and a subsequent one a coded frame overlap data acquisition unit and a second coding unit, wherein:
  • the coding mode switching identification unit after determining that the previous frame is encoded by the first coding mode, and the current frame needs to be encoded by using the second coding mode, triggering the previous coding frame overlapping data acquisition unit and The latter coded frame overlaps the data acquisition unit;
  • the previous coded frame overlap data acquiring unit obtains the coded data of the corresponding length in the corresponding previous frame according to the determined data length of the overlapped data of the current frame and the previous frame, for example, acquiring the previous frame.
  • the latter coded frame overlap data acquiring unit obtains the coded data of the corresponding length in the corresponding subsequent frame according to the determined data length of the coded data of the current frame and the subsequent frame, and provides the coded data to the transform domain.
  • a unit ie, a second coding unit; wherein, taking the TCX coding mode as the second coding mode as an example, considering that a corresponding smoothing scheme needs to be used in the coding process between the current TCX frames, in the device embodiment The unit can still be used for corresponding inter-frame smoothing;
  • the second coding unit performs overlapping processing on the coded data obtained by the previous coded frame overlap data acquisition unit and the subsequent coded frame overlap data acquisition unit and the coded data of the current frame to obtain a coding result, thereby implementing a frame Smoothing between.
  • the data length of the overlapped portion of the previous frame overlap data acquiring unit and the subsequent frame overlap data acquiring unit is determined according to the frame length of each coded frame; It is assumed that the data length of the overlapping portion used in the overlapping data acquiring unit of the previous frame is the first length, and the data length of the overlapping portion used in the data acquiring unit of the subsequent frame is the first The two lengths are the same between the first length and the second length, but are not limited to being the same.
  • the apparatus specifically includes a decoding mode switching identification unit, a previous decoding frame overlapping data acquiring unit, and a second decoding unit, where:
  • a decoding mode switching identification unit configured to: after further performing decoding operation on the received code stream, if it is further determined that the previous frame in the received code stream is decoded by using the first decoding mode, the current frame is used.
  • the second decoding mode is obtained by decoding, and the previous decoded frame overlapping data acquiring unit is triggered;
  • the previous decoded frame overlap data acquiring unit is configured to obtain, according to the determined data length of the decoded data of the current frame and the previous frame, the decoded data of the corresponding length in the corresponding previous frame, and provide the data to the first Two solutions Code unit
  • a second decoding unit configured to perform windowing overlap processing on the decoded data acquired by the previous decoded frame overlap data acquiring unit and the decoded data of the current frame to obtain a decoding result
  • a determining unit for overlapping the partial data length configured to determine, according to the indication information in the received code stream, the data length of the overlapping portion, and provide the previous decoded frame overlapping data acquiring unit, for example, Transmitting an encoding mode (ie, a frame length of the encoded frame) to the decoding end in the code stream, and after receiving the encoding mode, the decoding end determines the corresponding data length value of the overlapping portion according to the encoding mode, and of course, The overlapping portion data length value that the decoding end can use is indicated by other indication information.
  • an encoding mode ie, a frame length of the encoded frame
  • the first coding mode is a linear prediction coding mode
  • the second coding mode is a transform domain coding
  • the embodiments provided by the present invention are equally applicable to the problem caused by switching between two different encoding or decoding modes.
  • the overlap coding process can be performed on the two coding modes before and after, so that the coding and decoding quality loss caused by the handover is smoothed, and the coding and decoding quality is improved.
  • it may be applied to conversion smoothing processing from ACELP encoding mode to AAC (Advanced Audio Coding) mode, or to conversion smoothing processing from CELP (Linear Predictive Excitation Coding) mode to AAC mode, or , from ACELP coding mode to MDCT (Improved Discrete Cosine Transform) conversion smoothing between coding modes, and so on.

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Description

说明书 编解码方法及装置
[1] 技术领域
[2] 本发明涉及编解码技术领域, 尤其涉及一种编解码方法及装置。
[3] 发明背景
[4] 随着多媒体业务的逐渐广泛应用, 由于多媒体业务的自身特性使得在针对多媒 体业务的编码过程中需要更高的编码效率和实吋性, 同吋, 相应的音频的编码 带宽也需要进一步扩展。
[5] 目前, 业界釆用的能够满足低码率, 高质量音频编码的技术包括 AMR-WB+ (
Adaptive Multi-Rate Wideband
codec, 适应多速率的宽带多媒体信号编解码器) 技术。 基于 AMR-WB+的编码 器主要包括以下两种编码模式:
[6] (1) ACELP (Algebraic Code Excited Linear
Prediction, 代数码激励线性预测) 模式, 用于对语音进行编码;
[7] (2) TCX (Transform Coded
excitation, 变换码激励) 模式, 用于对乐音进行编码。
[8] 该 AMR-WB+技术是通过对低码率的语音编码进行扩展而形成, 具体是将针对 语音的 ACELP编码和针对乐音的 TCX编码结合在一起形成的一种混合编码方式 。 在对每帧进行编码过程中, 具体是通过比较两种模式的 SEGSNR (分段信噪比 ) 值来决定当前选择的编码模式。 其中, 一种模式切换情况为; 前一帧釆用的 是 ACELP编码模式, 当前帧需要釆用 TCX编码模式; 在这种情况下, 需要在编 码过程中釆用相应的策略以消除帧间的不连续性。 由于前一帧状态所得的零输 入响应和当前帧起始部分的信号有很大相似性, 为了使模式转换吋能保持良好 的平滑感, AMR-WB+在 ACELP编码帧到 TCX编码帧的转换吋使用了去除零输入 响应的实现方案。
[9] 对于前一帧釆用 ACELP编码模式, 当前帧釆用 TCX编码模式的模式切换情况, 相应的 TCX编码的实现过程如图 1所示, 在编码过程中, 输入的音频信号首先通 过感知加权滤波器后进行判断, 之后, 在感知加权后的信号中减去加窗处理后 的 ZIR (零输入响应) , 进而对获得的信号添加自适应窗、 进行变换域编码等处 理, 以获得相应的 TCX编码码流。
[10] 与图 1对应, 同样模式切换情况下, 相应的 TCX解码的实现过程如图 2所示, 输 入的 TCX编码的码流在通过变换域解码、 加窗并重叠相加后, 由于前一帧釆用 A CELP编码模式, 故在此需要给变换域解码的数据加上加窗处理后的 ZIR, 进而 再通过逆感知加权重建音频信号, 完成相应的 TCX解码操作。
[11] 在实现本发明过程中, 发明人发现: 在 AMR-WB+涉及的 TCX编解码过程中, 其釆用的消除帧间不连续的编解码方案的理论依据是零输入响应和当前帧起始 部分的信号相似, 因此, 当零输入响应和当前帧起始部分的信号不相似吋, 则 相应的消除帧间不连续的效果无法得到保证。 而且, 还由于在消除帧间不连续 的处理过程中, 需要计算合成加权滤波器的零输入响应, 且相应的算法复杂度 相对较高, 从而使得编解码过程的实现复杂程度也会相应提高。
[12] 发明内容
[13] 本发明的实施例提供了一种编解码方法及装置, 从而可以使得编解码过程中, 实现消除帧间不连续的处理过程的复杂程度降低, 进而降低了编解码过程的实 现复杂度。
[14] 本发明的实施例提供了一种编码方法, 包括:
[15] 当确定前一帧釆用第一编码模式进行编码, 且当前帧需要釆用第二编码模式进 行编码吋, 根据当前帧与前一帧的编码数据的交迭部分数据长度, 获取对应的 前一帧中的所述交迭部分数据长度的编码数据;
[16] 将从前一帧中获取的编码数据与当前帧的编码数据进行第二编码模式编码处理
, 获得编码结果。
[17] 本发明的实施例提供了一种编码装置, 包括编码模式切换识别单元、 前一编码 帧交迭数据获取单元和第二编码单元, 其中:
[18] 编码模式切换识别单元, 在确定前一帧釆用的是第一编码模式进行编码, 且当 前帧需要釆用变换域编码模式进行编码后, 触发所述前一编码帧交迭数据获取 单元; [19] 前一编码帧交迭数据获取单元, 用于根据确定的当前帧与前一帧的编码数据交 迭部分数据长度, 获取对应的前一帧中相应长度的编码数据, 并提供给第二编 码单元;
[20] 第二编码单元, 将所述前一编码帧交迭数据获取单元获取的编码数据与当前帧 的编码数据进行交迭处理, 获得编码结果。
[21] 本发明的实施例提供了一种解码方法, 包括:
[22] 对接收到的码流进行解码操作, 并确定前一帧釆用第一解码模式解码获得, 当 前帧为釆用第二解码模式解码获得;
[23] 根据确定的当前帧与前一帧的解码数据交迭部分数据长度, 获取对应的前一帧 中的所述交迭部分数据长度的解码数据;
[24] 将从前一帧中获取的解码数据与当前帧的解码数据进行交迭处理, 获得解码结 果。
[25] 本发明的实施例提供了一种解码装置, 包括解码模式切换识别单元、 前一解码 帧交迭数据获取单元和第二解码单元, 其中:
[26] 解码模式切换识别单元, 用于在根据解码后的码流中的信息确定前一帧釆用第 一解码模式解码获得, 当前帧为釆用第二码解码模式解码获得后, 触发前一解 码帧交迭数据获取单元;
[27] 前一解码帧交迭数据获取单元, 用于根据确定的当前帧与前一帧的解码数据交 迭部分数据长度, 获取对应的前一帧中相应长度的解码数据, 并提供给第二解 码单元;
[28] 第二解码单元, 用于将前一解码帧交迭数据获取单元获取的解码数据与当前帧 的解码数据进行交迭处理, 获得解码结果。
[29] 由上述本发明的实施例提供的技术方案可以看出, 本发明提供的实施例无需通 过滤波器运算便可以实现相应的模式切换编解码处理, 从而使得整个编解码过 程运算复杂度不高, 进而更易于软硬件的实现。 同吋, 本发明提供的实施例还 可以在零输入响应和当前帧起始部分的信号不相似情况下, 使得消除帧间不连 续的效果能够得到有效保证
[30] 附图简要说明 [31] 图 1为现有技术中 TCX编码过程的原理框图;
[32] 图 2为现有技术中 TCX解码过程的原理框图;
[33] 图 3为现有技术中计算加窗处理后的 ZIR值过程中应用的吋域窗函数
的示意图;
[34] 图 4为本发明实施例中 TCX编码过程的原理框图;
[35] 图 5为本发明实施例中 TCX解码过程的原理框图;
[36] 图 6为本发明实施例中输入语音帧结构示意图;
[37] 图 7为本发明实施例中加窗处理后的窗形示意图;
[38] 图 8为本发明实施例中解码过程中帧间交迭平滑的示意图;
[39] 图 9为本发明实施例中提供的编解码装置的具体实现结构示意图。
[40] 实施本发明的方式
[41] 本发明提供的编码的实施例包括: 在确定前一帧釆用的是第一编码模式进行编 码, 且进一步确定当前帧需要釆用第二编码模式进行编码, 即确定编码过程中 发生的编码模式的切换吋; 则根据当前帧与前一帧及后一帧的编码数据交迭部 分数据长度, 获取对应的前一帧及后一帧中的所述交迭部分数据长度的编码数 据; 并对从前一帧及后一帧数据中获取的编码数据及当前帧的编码数据进行基 于第二编码模式的编码处理, 获得编码结果。 其中, 所述的交迭部分数据长度 为根据各编码帧的帧长确定并预先设置于编码器中, 且所述编码帧的帧长越长 则对应的所述交迭部分数据长度也越长。
[42] 需要说明的是, 在该实施例中, 假设所述的当前帧与前一帧的编码数据交迭部 分数据长度为第一长度, 所述当前帧与后一帧的编码数据交迭部分数据长度为 第二长度, 则优选地, 所述第一长度可以与第二长度相同; 当然, 相应的实施 例在具体应用过程中并不限定为两长度值一定相同。
[43] 本发明提供的实施例中, 所述的第一编码模式具体可以但不限于为线性预测编 码模式, 所述的第二编码模式可以但不限于为变换域编码; 进一步, 相应的实 施例可以应用于各种线性预测编码与变换域编码之间的模式切换编码过程中, 例如, 由 ACELP编码到 TCX编码之间的模式切换, 等等。
[44] 对应的, 本发明提供的解码的实施例包括: 在对接收码流进行解码操作, 并确 定接收到的码流中的前一帧釆用第一解码模式解码获得, 当前帧为釆用第二解 码模式解码获得后, 则根据确定的当前帧与前一帧的解码数据对应的交迭部分 的数据长度, 获取对应的前一帧中的所述交迭部分数据长度的解码数据; 再将 从前一帧中获取的解码数据与当前帧的解码数据进行交迭处理, 具体为将前一 帧中的所述交迭部分数据长度的解码数据与当前帧的解码数据的头部进行加窗 叠加处理, 获得解码结果。
[45] 以 AMR-WB+编码为例, 针对由 ACELP编码 (即线性预测编码) 帧到 TCX编码
(即变换域编码) 帧的转换处理过程, 本发明的实施例提出了的一种 ACELP与 T CX编码模式切换的交迭平滑技术, 其能够在保证码率不变的情况下获得更好的 帧间平滑效果。 且在应用所述实施例过程中, 不需要通过复杂的合成感知加权 滤波器计算, 从而使得与现有技术中的 AMR-WB+的模式间平滑技术相比, 相应 的计算复杂度有所降低。
[46] 也就是说, 本发明的实施例主要是釆用模式间交迭平滑技术实现两种编码模式 之间切换吋对编码所造成的影响, 本实施例通过 TCX编解码处理, 以提高 TCX 编解码效率, 降低 TCX编解码复杂程度。 下面将对本发明提供的基于模式间交 迭平滑技术的 TCX编解码技术方案进行说明
[47] (一) 釆用模式间交迭平滑技术的 TCX编码方案
[48] 该方案的具体实现如图 4所示, 对于输入的需要进行 TCX编码操作的 TCX帧信 号, 具体是通过感知加权滤波器, 自适应加窗和变换域编码处理, 以获得到经 T CX编码后的码流; 其中, 如果前一帧釆用的是 ACELP模式编码, 则当前输入的 TCX帧信号将与后一帧交迭的数据长度减半, 同吋, 将省出的空间釆用前一帧 最后一个子帧的若干个釆样点值进行补足处理, 即分别釆用前一帧和后一帧的 交迭部分的编码数据与当前帧编码数据一起进行编码处理, 以实现帧间的平滑 处理。
[49] 可以看出, 在该图 4所示的实施例中, 无需再进行零输入响应的去除操作, 从 而可以简化编码过程中的复杂程度, 同吋, 由于当前帧与前、 后帧之间分别釆 用了交迭数据进行平滑操作, 从而可以实现帧间的有效平滑处理。
[50] (二) 釆用模式间交迭平滑技术的 TCX解码方案 [51] 与上述 TCX编码方案对应, 相应的 TCX解码方案的实现结构框图如图 5所示, 在解码处理过程中, TCX解码器接收到 TCX编码器发送来的经 TCX编码后的码 流, 之后通过变换域解码, 加窗和 TCX模式内的重叠相加, 然后再通过逆感知 加权滤波器得到合成音频信号; 其中, 若前一帧釆用的是 ACELP编码模式, 则 在解码端, 釆用与上述编码端对应的处理策略, 利用前一帧的解码结果中与当 前帧交迭部分进行交迭处理, 以获得当前帧的解码结果; 参照上述编码过程中 的实例, 则在 TCX解码器中, 将当前帧的起始交迭部分和上一帧 ACELP合成信 号的最后一子帧做加窗交迭, 以得到最终的合成音频信号。
[52] 为便于对本发明实施例的理解, 下面将结合附图对本发明实施例提供的编解码 算法做详细描述, 即对前一帧釆用 ACELP编码模式, 当前帧釆用 TCX编码模式 的情况下的编解码过程进行说明。
[53] (—) 编码过程
[54] 仍参照图 4所示, 对于前一帧使用 ACELP模式编码的情况, 且当前帧需要使用 TCX编码吋, 则具体可以釆用的帧间交迭技术包括:
[55] 根据当前帧 TCX编码模式 (如编码帧长分别为 256、 512或 1024的 TCX编码模式 ) , 将前一帧 ACELP处理过的最后若干个音频数据 (例如可以为 16、 32或 64点 s peech数据) 和当前帧音频数据一起进行 TCX编码, 所述的最后若干个音频数据 是指根据编码帧长确定的与前一帧交迭部分数据长度的音频数据;
[56] 相应的 TCX编码器输入的音频帧结构如图 6所示, 其中, L_frame (L帧) 表示 当前帧 TCX编码的帧长, 可以为 256、 512或 1024分别对应 TCX的三种编码模式 ; 其中, L1表示与前一帧交迭所取音频信号的长度, L2是为了与下一帧进行 ove rlap (交迭) 所取音频信号的釆样点数, L则表示当前帧所处理的实际音频信号 长度; 图 6中的各参数值具体可以为:
[57] 当1^_6¾11½=256吋, Ll=16, L2=16, L=288;
[58] 当 L_frame=512吋, Ll=32, L2=32, L=576;
[59] 当1^_6¾11½=1024吋, Ll=64, L2=64, L=1152。
[60] 因此, 当前帧与前一帧交迭的长度是随 TCX编码模式的变化而变化的, 具有自 适应的效果。 同吋, 此方法所处理 TCX每帧 speech (话音) 信号的实际帧长和 A MR-WB+中的实际帧长匹配, 保证了编码的正确性。
[61] 所述的需要进行 TCX编码操作的 speech信号通过感知加权滤波器处理后, 进行 自适应加窗操作的窗形参照图 7所示, 其中:
[62] w
Figure imgf000009_0001
, 其中, n=L2,..., (2L2-1) ;
[63] 其中, w(n)是指图 7中 L2所示部分的曲线; 也就是说, 对前一帧交迭的部分没 有加窗, 而对与后一帧的交迭的部分加余弦窗 w(n) ;
[64] 而且, 因为设置了与前一帧的交迭部分, 因此, 该余弦窗的窗长仅是 AMR-WB
+中的余弦窗窗长的一半。
[65] 另外, 当下一帧还是 TCX编码吋, 则下一帧帧头所加的窗长要和 L2长度一致, 即相应的交迭部分长度应和当前帧一致, 以保证帧间的平滑效果。
[66] (二) 解码过程
[67] 与上述编码过程对应, TCX解码器从接收到的当前帧 TCX编码码流中, 解码得 到当前帧的合成音频信号, 将其头部交迭部分与前一帧 ACELP的解码的音频信 号加窗做叠加, 从而得到最终的合成音频输出。
[68] 具体为: 将前一帧 ACELP编码解出的合成音频信号和当前帧 TCX编码所解码出 的音频信号进行如图 8所示的加窗操作, 然后, 将交迭部分进行叠加得到最后的 合成音频信号。
[69] 参照图 8所示, 在交迭部分釆用了三角窗, 对 ACELP最后 L1个样点的合成音频 信号釆用 w2(n)表示, 对 TCX交迭部分的合成音频信号釆用 wl(n)表示, 则相应的 合成音频信号具体如下式所示:
[70] (1) wl(n)=n/Ll , for n = 0,...,L1 ;
[71] (2) w2(n)=(Ll-n)/Ll , for n = 0,...,L1 ;
[72] 经过上述过程, 便可以顺利地完成相应的 TCX解码操作, 以获得相应的 TCX解 码操作结果。
[73] 本发明还提供了一种编解码装置的实施例, 具体如图 9所示, 具体包括编码装 置及解码装置, 下面将分别对该两装置的具体实现结构进行描述。
[74] (一) 编码装置
[75] 该装置具体包括编码模式切换识别单元、 前一编码帧交迭数据获取单元、 后一 编码帧交迭数据获取单元和第二编码单元, 其中:
[76] 编码模式切换识别单元, 在确定前一帧釆用第一编码模式进行编码, 且当前帧 需要釆用第二编码模式进行编码后, 触发所述前一编码帧交迭数据获取单元及 后一编码帧交迭数据获取单元;
[77] 前一编码帧交迭数据获取单元, 根据确定的当前帧与前一帧的编码数据交迭部 分数据长度, 获取对应的前一帧中相应长度的编码数据, 例如, 获取前一帧中 的相应长度的编码数据, 并提供给变换域编码单元 (即第二编码单元) ;
[78] 后一编码帧交迭数据获取单元, 根据确定的当前帧与后一帧的编码数据交迭部 分数据长度, 获取对应的后一帧中相应长度的编码数据, 并提供给变换域编码 单元 (即第二编码单元) ; 其中, 以 TCX编码模式作为第二编码模式为例, 考 虑到目前 TCX帧之间的编码过程中需要釆用相应的平滑处理方案, 故在该装置 实施例中可以仍釆用该单元, 以进行相应的帧间平滑处理;
[79] 第二编码单元, 将前一编码帧交迭数据获取单元及后一编码帧交迭数据获取单 元获取的编码数据与当前帧的编码数据进行交迭处理, 获得编码结果, 从而实 现帧间的平滑处理。
[80] 在该装置中, 所述的前一帧交迭数据获取单元及后一帧交迭数据获取单元中釆 用的交迭部分数据长度为根据各编码帧的帧长预先确定; 具体一点讲, 假设所 述的前一帧交迭数据获取单元中釆用的交迭部分数据长度为第一长度, 与所述 后一帧交迭数据获取单元中釆用的交迭部分数据长度为第二长度, 则所述第一 长度与第二长度之间相同, 但不限于一定相同。
[81] (二) 解码装置
[82] 该装置具体包括解码模式切换识别单元、 前一解码帧交迭数据获取单元和第二 解码单元, 其中:
[83] 解码模式切换识别单元, 用于在对接收到的码流进行解码操作后, 若进一步确 定收到的码流中的前一帧釆用第一解码模式解码获得, 当前帧为釆用第二解码 模式解码获得, 则触发所述前一解码帧交迭数据获取单元;
[84] 前一解码帧交迭数据获取单元, 用于根据确定的当前帧与前一帧的解码数据交 迭部分数据长度, 获取对应的前一帧中相应长度的解码数据, 并提供给第二解 码单元;
[85] 第二解码单元, 用于将前一解码帧交迭数据获取单元获取的解码数据与当前帧 的解码数据进行加窗交迭处理, 获得解码结果;
[86] 交迭部分数据长度的确定单元, 用于根据收到的码流中的指示信息确定所述交 迭部分数据长度, 并提供给所述前一解码帧交迭数据获取单元, 例如, 在码流 中将编码模式 (即编码帧的帧长) 传递给解码端, 解码端收到所述编码模式后 则根据该编码模式确定对应的所述交迭部分数据长度值, 当然, 也可以通过其 他指示信息指示解码端可以釆用的交迭部分数据长度值。
[87] 在上述装置中, 所述的第一编码模式为线性预测编码模式, 所述的第二编码模 式为变换域编码。
[88] 需要说明的是, 本发明提供的各实施例同样适用于两种不同编码或解码模式之 间进行切换吋造成的问题。 具体可以通过对前后两种编码模式进行交叠编码处 理, 从而平滑由于切换带来的编码解码质量损失, 提高编解码质量。 例如, 具 体还可以应用于由 ACELP编码模式到 AAC (高级音频编码) 模式之间的转换平 滑处理, 或者, 应用于由 CELP (线性预测激励编码) 模式到 AAC模式之间的转 换平滑处理, 或者, 由 ACELP编码模式到 MDCT (改进的离散余弦变换) 编码 模式之间的转换平滑处理, 等等。
[89] 综上所述, 由于在解码端合成音频信号上直接做交迭运算, 因而可以获得较好 的帧间平滑效果。 而且, 本发明提供的实施例无需通过滤波器运算, 从而使得 整个编解码过程运算复杂度不高, 更易于软硬件的实现。
[90] 以上所述, 仅为本发明较佳的具体实施方式, 但本发明的保护范围并不局限于 此, 任何熟悉本技术领域的技术人员在本发明揭露的技术范围内, 可轻易想到 的变化或替换, 都应涵盖在本发明的保护范围之内。 因此, 本发明的保护范围 应该以权利要求的保护范围为准。

Claims

权利要求书
[1] 1、 一种编码方法, 其特征在于, 该方法包括:
当前一帧釆用第一编码模式进行编码, 且当前帧需要釆用第二编码模式进 行编码吋, 根据当前帧与前一帧编码数据的交迭部分数据长度, 获取对应 的前一帧中的所述交迭部分数据长度的编码数据;
对获取的前一帧中的所述交迭部分数据长度的编码数据和当前帧的编码数 据进行第二编码模式编码处理, 获得编码结果。
[2] 2、 根据权利要求 1所述的方法, 其特征在于, 所述的第一编码模式为线性 预测编码模式, 所述的第二编码模式为变换域编码。
[3] 3、 根据权利要求 1所述的方法, 其特征在于, 所述的交迭部分数据长度的 确定方式包括: 根据各编码帧的帧长确定对应的所述交迭部分数据长度。
[4] 4、 根据权利要求 1、 2或 3所述的方法, 其特征在于, 所述的方法还包括: 根据确定当前帧与后一帧的编码数据交迭部分数据长度获取后一帧中的编 码数据, 并将获取的编码数据与从前一帧中获取的编码数据及当前帧的编 码数据进行变换域编码处理。
[5] 5、 根据权利要求 4所述的方法, 其特征在于, 所述的当前帧与前一帧的编 码数据交迭部分数据长度, 和所述当前帧与后一帧的编码数据交迭部分数 据长度相同。
[6] 6、 一种编码装置, 其特征在于, 包括编码模式切换识别单元、 前一编码帧 交迭数据获取单元和第二编码单元, 其中:
编码模式切换识别单元, 用于在确定前一帧釆用的第一编码模式进行编码 , 且当前帧需要釆用第二编码模式进行编码后, 触发所述前一编码帧交迭 数据获取单元;
前一编码帧交迭数据获取单元, 用于根据当前帧与前一帧的编码数据交迭 部分数据长度, 获取对应的前一帧中相应长度的编码数据; 第二编码单元, 用于对所述前一编码帧交迭数据获取单元获取的编码数据 及当前帧的编码数据进行第二编码模式编码处理, 获得编码结果。
[7] 7、 根据权利要求 6所述的装置, 其特征在于, 所述的第一编码模式为线性 预测编码模式, 所述的第二编码模式为变换域编码。
[8] 8、 根据权利要求 6所述的装置, 其特征在于, 所述的前一帧交迭数据获取 单元及后一帧交迭数据获取单元中釆用的交迭部分数据长度为根据各编码 帧的帧长预先确定。
[9] 9、 根据权利要求 6、 7或 8所述的装置, 其特征在于, 所述的方法还包括后 一编码帧交迭数据获取单元, 用于根据确定的当前帧与后一帧的编码数据 交迭部分数据长度, 获取对应的后一帧中相应长度的编码数据, 并提供给 变换域编码单元进行编码操作。
[10] 10、 根据权利要求 9所述的装置, 其特征在于, 所述的前一帧交迭数据获取 单元中釆用的交迭部分数据长度, 与所述后一帧交迭数据获取单元中釆用 的交迭部分数据长度相同。
[11] 11、 一种解码方法, 其特征在于, 包括:
对接收到的码流进行解码操作, 并确定前一帧釆用第一解码模式解码获得
, 当前帧为釆用第二解码模式解码获得;
根据确定的当前帧与前一帧的解码数据交迭部分数据长度, 获取对应的前 一帧中的所述交迭部分数据长度的解码数据;
将从前一帧中获取的解码数据与当前帧的解码数据进行交迭处理, 获得解 码结果。
[12] 12、 根据权利要求 11所述的方法, 其特征在于, 所述的第一编码模式为线 性预测编码模式, 所述的第二编码模式为变换域编码。
[13] 13、 根据权利要求 11所述的方法, 其特征在于, 所述的交迭部分数据长度 的确定方式包括: 根据收到的码流中的指示信息确定所述交迭部分数据长 度。
[14] 14、 一种解码装置, 其特征在于, 包括解码模式切换识别单元、 前一解码 帧交迭数据获取单元和第二解码单元, 其中:
解码模式切换识别单元, 用于在根据解码后的码流中的信息确定前一帧为 釆用第一解码模式解码获得, 当前帧为釆用第二解码模式解码获得后, 触 发前一解码帧交迭数据获取单元; 前一解码帧交迭数据获取单元, 用于根据当前帧与前一帧的解码数据交迭 部分数据长度, 获取对应的前一帧中相应长度的解码数据, 并提供给第二 解码单元;
第二解码单元, 用于将前一解码帧交迭数据获取单元获取的解码数据与当 前帧的解码数据进行交迭处理, 获得解码结果。
[15] 15、 根据权利要求 14所述的装置, 其特征在于, 所述的第一编码模式为线 性预测编码模式, 所述的第二编码模式为变换域编码。
[16] 16、 根据权利要求 14或 15所述的装置, 其特征在于, 所述的装置还包括交 迭部分数据长度的确定单元, 用于根据收到的码流中的指示信息确定所述 交迭部分数据长度。
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Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2936898A1 (fr) * 2008-10-08 2010-04-09 France Telecom Codage a echantillonnage critique avec codeur predictif
US20110153333A1 (en) * 2009-06-23 2011-06-23 Bruno Bessette Forward Time-Domain Aliasing Cancellation with Application in Weighted or Original Signal Domain
JP2013507648A (ja) * 2009-10-08 2013-03-04 フラウンホッファー−ゲゼルシャフト ツァ フェルダールング デァ アンゲヴァンテン フォアシュンク エー.ファオ 線形予測符号化ベースのノイズ整形を用いた多重モードオーディオ信号デコーダ、多重モードオーディオ信号エンコーダ、方法およびコンピュータプログラム
US9093066B2 (en) 2010-01-13 2015-07-28 Voiceage Corporation Forward time-domain aliasing cancellation using linear-predictive filtering to cancel time reversed and zero input responses of adjacent frames

Families Citing this family (25)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2911228A1 (fr) * 2007-01-05 2008-07-11 France Telecom Codage par transformee, utilisant des fenetres de ponderation et a faible retard.
KR101649376B1 (ko) 2008-10-13 2016-08-31 한국전자통신연구원 Mdct 기반 음성/오디오 통합 부호화기의 lpc 잔차신호 부호화/복호화 장치
WO2010044593A2 (ko) 2008-10-13 2010-04-22 한국전자통신연구원 Mdct 기반 음성/오디오 통합 부호화기의 lpc 잔차신호 부호화/복호화 장치
US9384748B2 (en) 2008-11-26 2016-07-05 Electronics And Telecommunications Research Institute Unified Speech/Audio Codec (USAC) processing windows sequence based mode switching
KR101315617B1 (ko) * 2008-11-26 2013-10-08 광운대학교 산학협력단 모드 스위칭에 기초하여 윈도우 시퀀스를 처리하는 통합 음성/오디오 부/복호화기
KR101622950B1 (ko) * 2009-01-28 2016-05-23 삼성전자주식회사 오디오 신호의 부호화 및 복호화 방법 및 그 장치
KR101411759B1 (ko) * 2009-10-20 2014-06-25 프라운호퍼 게젤샤프트 쭈르 푀르데룽 데어 안겐반텐 포르슝 에. 베. 오디오 신호 인코더, 오디오 신호 디코더, 앨리어싱-소거를 이용하여 오디오 신호를 인코딩 또는 디코딩하는 방법
BR122022013454B1 (pt) 2009-10-20 2023-05-16 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Codificador de áudio, decodificador de áudio, método para codificar uma informação de áudio, método para decodificar uma informação de áudio que utiliza uma detecção de um grupo de valores espectrais previamente decodificados
AU2011206677B9 (en) 2010-01-12 2014-12-11 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Audio encoder, audio decoder, method for encoding and decoding an audio information, and computer program obtaining a context sub-region value on the basis of a norm of previously decoded spectral values
CN102934161B (zh) * 2010-06-14 2015-08-26 松下电器产业株式会社 音频混合编码装置以及音频混合解码装置
EP3079153B1 (en) 2010-07-02 2018-08-01 Dolby International AB Audio decoding with selective post filtering
PL4372742T3 (pl) 2010-07-08 2026-02-02 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Koder wykorzystujący kasowanie aliasingu w przód
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EP2980797A1 (en) 2014-07-28 2016-02-03 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Audio decoder, method and computer program using a zero-input-response to obtain a smooth transition
EP2980796A1 (en) 2014-07-28 2016-02-03 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Method and apparatus for processing an audio signal, audio decoder, and audio encoder
CN106816153B (zh) * 2015-12-01 2019-03-15 腾讯科技(深圳)有限公司 一种数据处理方法及其终端
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CN115346537B (zh) * 2021-05-14 2024-11-29 华为技术有限公司 一种音频编码、解码方法及装置
CN118522296A (zh) * 2023-02-17 2024-08-20 华为技术有限公司 有损编解码器和无损编解码器之间的切换方法和装置

Citations (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1212562A (zh) * 1997-07-04 1999-03-31 松下电器产业株式会社 图像预测解码方法、解码装置、编码装置及数据存储媒体
CN1338104A (zh) * 1999-01-28 2002-02-27 多尔拜实验特许公司 自适应块长编码系统的数据分帧
WO2002049008A1 (en) * 2000-12-15 2002-06-20 British Telecommunications Public Limited Company Encoding audio signals
US6556624B1 (en) * 1999-07-27 2003-04-29 At&T Corp. Method and apparatus for accomplishing multiple description coding for video
CN1461563A (zh) * 2001-04-10 2003-12-10 皇家菲利浦电子有限公司 帧序列的编码方法
CN1615659A (zh) * 2002-01-18 2005-05-11 皇家飞利浦电子股份有限公司 音频编码
CN1669075A (zh) * 2002-07-16 2005-09-14 皇家飞利浦电子股份有限公司 音频编码
JP2006330256A (ja) * 2005-05-25 2006-12-07 Kddi Corp オーディオ信号に対する電子透かし埋込み方法および検出方法

Family Cites Families (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6134518A (en) * 1997-03-04 2000-10-17 International Business Machines Corporation Digital audio signal coding using a CELP coder and a transform coder
US6233550B1 (en) * 1997-08-29 2001-05-15 The Regents Of The University Of California Method and apparatus for hybrid coding of speech at 4kbps
JP3806770B2 (ja) * 2000-03-17 2006-08-09 松下電器産業株式会社 窓処理装置および窓処理方法
CA2457988A1 (en) * 2004-02-18 2005-08-18 Voiceage Corporation Methods and devices for audio compression based on acelp/tcx coding and multi-rate lattice vector quantization
US7596486B2 (en) * 2004-05-19 2009-09-29 Nokia Corporation Encoding an audio signal using different audio coder modes

Patent Citations (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1212562A (zh) * 1997-07-04 1999-03-31 松下电器产业株式会社 图像预测解码方法、解码装置、编码装置及数据存储媒体
CN1338104A (zh) * 1999-01-28 2002-02-27 多尔拜实验特许公司 自适应块长编码系统的数据分帧
US6556624B1 (en) * 1999-07-27 2003-04-29 At&T Corp. Method and apparatus for accomplishing multiple description coding for video
WO2002049008A1 (en) * 2000-12-15 2002-06-20 British Telecommunications Public Limited Company Encoding audio signals
CN1461563A (zh) * 2001-04-10 2003-12-10 皇家菲利浦电子有限公司 帧序列的编码方法
CN1615659A (zh) * 2002-01-18 2005-05-11 皇家飞利浦电子股份有限公司 音频编码
CN1669075A (zh) * 2002-07-16 2005-09-14 皇家飞利浦电子股份有限公司 音频编码
JP2006330256A (ja) * 2005-05-25 2006-12-07 Kddi Corp オーディオ信号に対する電子透かし埋込み方法および検出方法

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
See also references of EP2120233A4 *

Cited By (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2936898A1 (fr) * 2008-10-08 2010-04-09 France Telecom Codage a echantillonnage critique avec codeur predictif
WO2010040937A1 (fr) * 2008-10-08 2010-04-15 France Telecom Codage a echantillonnage critique avec codeur predictif
US8880411B2 (en) 2008-10-08 2014-11-04 Orange Critical sampling encoding with a predictive encoder
US20110153333A1 (en) * 2009-06-23 2011-06-23 Bruno Bessette Forward Time-Domain Aliasing Cancellation with Application in Weighted or Original Signal Domain
US8725503B2 (en) * 2009-06-23 2014-05-13 Voiceage Corporation Forward time-domain aliasing cancellation with application in weighted or original signal domain
JP2013507648A (ja) * 2009-10-08 2013-03-04 フラウンホッファー−ゲゼルシャフト ツァ フェルダールング デァ アンゲヴァンテン フォアシュンク エー.ファオ 線形予測符号化ベースのノイズ整形を用いた多重モードオーディオ信号デコーダ、多重モードオーディオ信号エンコーダ、方法およびコンピュータプログラム
US9093066B2 (en) 2010-01-13 2015-07-28 Voiceage Corporation Forward time-domain aliasing cancellation using linear-predictive filtering to cancel time reversed and zero input responses of adjacent frames

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