公开(公告)号：WO2009029035A1

公开(公告)日：2009-03-05

申请号：PCT/SE2008/050967

申请日：2008-08-26

Applicant: TELEFONAKTIEBOLAGET LM ERICSSON (PUBL) ,  BRIAND, Manuel ,  TALEB, Anisse

Inventor： BRIAND, Manuel ,  TALEB, Anisse

IPC: G10L19/02 ,  H04B1/66

CPC classification number: G10L19/0204 ,  G10L19/0212 ,  G10L19/035

Abstract: In a method of perceptual transform coding of audio signals in a telecommunication system, performing the steps of determining transform coefficients representative of a time to frequency transformation of a time segmented input audio signal; determining a spectrum of perceptual sub-bands for said input audio signal based on said determined transform coefficients; determining masking thresholds for each said sub-band based on said determined spectrum; computing scale factors for each said sub-band based on said determined masking thresholds, and finally adapting said computed scale factors for each said sub-band to prevent energy loss for perceptually relevant sub-bands.

Abstract translation: 在电信系统中对音频信号的感知变换编码的方法中，执行以下步骤：确定表示时间分段输入音频信号的时间到频率变换的变换系数; 基于所述确定的变换系数确定所述输入音频信号的感知子带的频谱; 基于所述确定的频谱确定每个所述子带的掩蔽阈值; 基于所述确定的掩蔽阈值计算每个所述子带的比例因子，并且最后适应每个所述子带的所述计算的比例因子以防止感知相关子带的能量损失。

公开(公告)号：WO2007078227A1

公开(公告)日：2007-07-12

申请号：PCT/SE2006/001366

申请日：2006-11-29

Applicant: TELEFONAKTIEBOLAGET LM ERICSSON (PUBL) ,  TALEB, Anisse ,  FRÖJDH, Per

Inventor： TALEB, Anisse ,  FRÖJDH, Per

IPC: H04N7/52

CPC classification number: H04N21/4341 ,  H04N21/2335 ,  H04N21/234327 ,  H04N21/23439 ,  H04N21/2368 ,  H04N21/2385 ,  H04N21/2402 ,  H04N21/2662 ,  H04N21/85406

Abstract: The invention involves collective management of video (30) and audio (40) content in a content provider (100). The video (30) and audio (40) content is available in multiple potential versions (32, 34, 36; 42, 44, 46), e.g. in the form of scalable media (36, 46) or media (32, 34; 42, 44) pre-encoded to fixed bandwidth levels. The video (30) and audio (40) data is associated with bandwidth share information (62, 64, 66) enabling estimation of a respective apportionment of a total available bandwidth to the video (30) and audio (40) content. The provider (100) uses this share information (62, 64, 66) and information of the total assignable bandwidth level for providing a respective video version (32, 34, 36) and audio version (42, 44, 46) from the multiple potential versions (32, 34, 36, 42, 44, 46). This allows for increased user-quality when rendering the video (30) and audio (40) data as optimal video (32, 34, 36) and audio (42, 44, 46) version can be dynamically provided during the media session.

Abstract translation: 本发明涉及对内容提供商（100）中的视频（30）和音频（40）内容的集体管理。 视频（30）和音频（40）内容可在多个潜在版本（32,34,36; 42,44,46）中获得，例如， 以预编码到固定带宽水平的可伸缩媒体（36,46）或媒体（32,34; 42,44）的形式。 视频（30）和音频（40）数据与带宽共享信息（62,64,66）相关联，使得能够估计对视频（30）和音频（40）内容的总可用带宽的相应分配。 提供者（100）使用该共享信息（62,64,66）和总可分配带宽级别的信息，以从多个网络提供相应的视频版本（32,34,36）和音频版本（42,44,46） 潜在版本（32,34,36,42,44,46）。 这可以在将视频（30）和音频（40）数据呈现为最佳视频（32,34,36）时提高用户质量，并且可以在媒体会话期间动态地提供音频（42,44,46）版本。

公开(公告)号：WO2007073328A3

公开(公告)日：2007-06-28

申请号：PCT/SE2006/050560

申请日：2006-12-07

Applicant: TELEFONAKTIEBOLAGET LM ERICSSON (publ) ,  TALEB, Anisse ,  JANSSON, Fredrik ,  HÅKANSSON, Stefan ,  JOHANSSON, Ingemar ,  BRUHN, Stefan ,  KIM, Moo Young

Inventor： TALEB, Anisse ,  JANSSON, Fredrik ,  HÅKANSSON, Stefan ,  JOHANSSON, Ingemar ,  BRUHN, Stefan ,  KIM, Moo Young

IPC: H04L12/66 ,  H04L1/18 ,  H04L12/56

Abstract: A network processing node (e.g., MGW, MRFP) and method are described herein that can: (1) receive packets on a first heterogeneous link (e.g., wireless link); (2) manipulate the received packets based on known characteristics about a second heterogeneous link (e.g., "Internet" link); and (3) send the manipulated packets on the second heterogeneous link (e.g., "Internet" link). For example, the network processing node can manipulate the received packets by adding redundancy, removing redundancy, frame aggregating (re-packetizing), recovering lost packets and/or re-transmitting packets.

公开(公告)号：WO2006091139A1

公开(公告)日：2006-08-31

申请号：PCT/SE2005/002033

申请日：2005-12-22

Applicant: TELEFONAKTIEBOLAGET LM ERICSSON (PUBL) ,  TALEB, Anisse ,  ANDERSSON, Stefan

Inventor： TALEB, Anisse ,  ANDERSSON, Stefan

IPC: H04B1/66 ,  G10L19/00

CPC classification number: G10L19/022 ,  G10L19/002 ,  G10L19/008 ,  G10L19/24 ,  G10L19/26

Abstract: The invention provides a highly efficient technique for encoding a multi-channel audio signal. The invention relies on the basic principle of encoding a first signal representation of one or more of the multiple channels in a first encoder (130) and encoding a second signal representation of one or more of the multiple channels in a second, multi-stage, encoder (140). This procedure is significantly enhanced by providing a controller (150) for adaptively allocating a number of encoding bits among the different encoding stages of the second, multi-stage, encoder (140) in dependence on multi-channel audio signal characteristics.

Abstract translation: 本发明提供了一种用于编码多声道音频信号的高效技术。 本发明依赖于在第一编码器（130）中编码多个信道中的一个或多个信道的第一信号表示的基本原理，并且在第二多级信道中编码多个信道中的一个或多个信道的第二信号表示， 编码器（140）。 通过提供一种用于根据多声道音频信号特性在第二，多级编码器（140）的不同编码级之间自适应地分配多个编码位的控制器（150）来显着增强该过程。

公开(公告)号：WO2010042024A1

公开(公告)日：2010-04-15

申请号：PCT/SE2009/051071

申请日：2009-09-25

Applicant: TELEFONAKTIEBOLAGET LM ERICSSON (PUBL) ,  NORVELL, Erik ,  SEHLSTEDT, Martin ,  TALEB, Anisse

Inventor： NORVELL, Erik ,  SEHLSTEDT, Martin ,  TALEB, Anisse

IPC: G10L19/00

CPC classification number: G10L19/008

Abstract: The invention relates to the technical field of audio encoding and/or decoding technologies, and thus concerns an overall encoding procedure and associated decoding procedure. The encoding procedure involves at least two signal encoding processes (S1-S3) operating on signal representations of a set of audio input channels, as well as residual encoding (S7-S8). It also involves a dedicated process (S4-S6) to estimate and encode energies of the audio input channels. Each encoding process is associated with a corresponding decoding process. In the overall decoding procedure the decoded signals from each encoding process are preferably combined such that the output channels are close to the input channels in terms of energy and/or quality. Normally, the combination step also adapts to the possible loss of one or more signal representation in part or in whole, such that the energy and quality is optimized with the signals at hand in the decoder. In this way, the overall quality of the output channels is improved.

Abstract translation: 本发明涉及音频编码和/或解码技术的技术领域，因此涉及整体编码过程和相关联的解码过程。 编码过程涉及对一组音频输入通道的信号表示进行操作的至少两个信号编码处理（S1-S3）以及残差编码（S7-S8）。 它还涉及专门的过程（S4-S6）来估计和编码音频输入通道的能量。 每个编码过程与相应的解码过程相关联。 在整个解码过程中，优选地组合来自每个编码处理的解码信号，使得输出信道在能量和/或质量方面靠近输入信道。 通常，组合步骤还适应于部分或全部的一个或多个信号表示的可能损失，使得能量和质量通过解码器中的手头信号被优化。 这样，输出通道的整体质量得到提高。

公开(公告)号：WO2009029032A3

公开(公告)日：2009-03-05

申请号：PCT/SE2008/050959

申请日：2008-08-25

Applicant: TELEFONAKTIEBOLAGET LM ERICSSON (PUBL) ,  TALEB, Anisse

Inventor： TALEB, Anisse

IPC: G10L19/02

Abstract: The signal processing is based on the c oncept of using a time-domain aliased (12, TDA) frame as a basis for time segmen tation (14) and spectral analysis (16), performing segmentation in time based on the time-domain aliased frame and performing spectral analysis based on the resulting time segments. The time resolution of the overall ?segmented? time-to-frequenc y transform can thus be changed by simply adapting the time segmentation to ob tain a suitable number of time segments based on which spectral analysis is applied. The overall set of spectral coefficients, obtained for all the segments, provides a selectable time-frequency tiling of the original signal frame.

公开(公告)号：WO2007035148A2

公开(公告)日：2007-03-29

申请号：PCT/SE2006/001043

申请日：2006-09-12

Applicant: TELEFONAKTIEBOLAGET LM ERICSSON (publ) ,  TALEB, Anisse

Inventor： TALEB, Anisse

IPC: H03M7/30

CPC classification number: H03M7/3082

Abstract: A vector quantizer includes a lattice quantizer (10) approximating a vector x by a lattice vector belonging to a lattice Λ 0 . A lattice vector decomposer (14) connected to the lattice quantizer successively decomposes the lattice vector into a sequence of quotient vectors y, and a sequence of remainder vectors r i on successive lattices Λ I-1 by lattice division with a corresponding predetermined sequence of integers p i ≥ 2 , where i = l...k and k is a positive integer representing the number of elements in each sequence.

Abstract translation: 矢量量化器包括通过属于格子φ0的晶格矢量近似矢量x的晶格量化器（10）。 连接到晶格量化器的晶格矢量分解器（14）将晶格矢量依次分解成一个商矢量y的序列，并且在连续晶格上的剩余矢量序列ΠI-1 通过与相应的预定的整数序列p i i进行点划分

公开(公告)号：WO2006091151B1

公开(公告)日：2006-12-14

申请号：PCT/SE2006000235

申请日：2006-02-22

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE ,  ANDERSSON STEFAN

Inventor： TALEB ANISSE ,  ANDERSSON STEFAN

IPC: G10L19/00 ,  H04B1/66

CPC classification number: G10L19/008 ,  G10L19/24

Abstract: The invention provides an efficient technique for encoding a multi-channel audio signal. The invention relies on the principle of encoding (Sl) a signal representation of one or more of the multiple channels in a first encoding process, and encoding another signal representation of one or more channels in a second, filter-based encoding process. A basic idea according to the invention is to select (S2), for the second encoding process, a combination of i) frame division configuration of an overall encoding frame into a set of sub-frames, and ii) filter length for each sub-frame, according to a predetermined criterion. The second signal representation is then encoded (S3) in each sub-frame of the overall encoding frame according to the selected combination. The possibility to select frame division configuration and at the same time adjust the filter length for each sub-frame provides added degrees of freedom, and generally results in improved performance.

Abstract translation: 本发明提供了一种用于对多声道音频信号进行编码的有效技术。 本发明依赖于在第一编码过程中编码（S1）一个或多个多个信道的信号表示的原理，以及在第二个基于过滤器的编码过程中编码一个或多个信道的另一个信号表示。 根据本发明的基本思想是，对于第二编码处理，选择（S2）i）将整个编码帧的帧分配配置成一组子帧的组合，以及ii）每个子帧的滤波器长度， 帧，根据预定标准。 然后根据所选择的组合，在整个编码帧的每个子帧中对第二信号表示进行编码（S3）。 选择帧分配配置并同时调整每个子帧的滤波器长度的可​​能性提供了附加的自由度，并且通常导致改进的性能。

公开(公告)号：WO2006091151A1

公开(公告)日：2006-08-31

申请号：PCT/SE2006/000235

申请日：2006-02-22

Applicant: TELEFONAKTIEBOLAGET LM ERICSSON ,  TALEB, Anisse ,  ANDERSSON, Stefan

Inventor： TALEB, Anisse ,  ANDERSSON, Stefan

IPC: G10L19/00 ,  H04B1/66

CPC classification number: G10L19/008 ,  G10L19/24

Abstract: The invention provides an efficient technique for encoding a multi-channel audio signal. The invention relies on the principle of encoding (Sl) a signal representation of one or more of the multiple channels in a first encoding process, and encoding another signal representation of one or more channels in a second, filter-based encoding process. A basic idea according to the invention is to select (S2), for the second encoding process, a combination of i) frame division configuration of an overall encoding frame into a set of sub-frames, and ii) filter length for each sub-frame, according to a predetermined criterion. The second signal representation is then encoded (S3) in each sub-frame of the overall encoding frame according to the selected combination. The possibility to select frame division configuration and at the same time adjust the filter length for each sub-frame provides added degrees of freedom, and generally results in improved performance.

Abstract translation: 本发明提供了用于编码多声道音频信号的有效技术。 本发明依赖于在第一编码过程中对多个信道中的一个或多个信道的信号表示进行编码（S1）以及在第二基于过滤器的编码过程中对一个或多个信道的另一信号表示进行编码的原理。 根据本发明的基本思想是针对第二编码处理选择（S2）i）全部编码帧的帧划分配置到一组子帧中的组合，以及ii）针对每个子帧的滤波器长度， 帧，根据预定的标准。 然后根据所选择的组合，在整个编码帧的每个子帧中对第二信号表示进行编码（S3）。 选择帧划分配置并同时调整每个子帧的滤波器长度的可​​能性提供了增加的自由度，并且通常导致改进的性能。

公开(公告)号：WO2009038512A1

公开(公告)日：2009-03-26

申请号：PCT/SE2008/000272

申请日：2008-04-17

Applicant: TELEFONAKTIEBOLAGET LM ERICSSON (PUBL) ,  NORVELL, Erik ,  TALEB, Anisse

Inventor： NORVELL, Erik ,  TALEB, Anisse

IPC: G10L19/00 ,  H04S3/00

CPC classification number: G10L19/24 ,  G10L19/008

Abstract: An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes (Sl, S4) operating on signal representations of a set of audio input channels. Local synthesis (S2) is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input (S3) to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals (S5) from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding (S6) in a further encoding process, preferably based on correlation between the residual error signals.

Abstract translation: 提出了一种整体编码过程和相关的解码过程。 编码过程涉及对一组音频输入通道的信号表示进行操作的至少两个信号编码处理（S1，S4）。 本地合成（S2）与第一编码处理结合使用以产生本地解码的信号，包括第一编码处理的编码误差的表示。 该本地解码信号作为输入（S3）应用于第二编码处理。 整个编码过程从至少一个所述编码过程产生至少两个残留编码误差信号（S5），包括至少所述第二编码处理。 然后，在进一步的编码处理中，优选地基于残差误差信号之间的相关性，对剩余误差信号进行复合残差编码（S6）。