公开(公告)号：WO01022403A1

公开(公告)日：2001-03-29

申请号：PCT/US2000/025869

申请日：2000-09-20

IPC: G10L19/02 ,  G10L19/00 ,  G10L19/04 ,  G10L19/08 ,  G10L19/14

CPC classification number: G10L19/173 ,  G10L19/087

Abstract: An enhanced low-bit rate parametric voice coder that groups a number of frames from an underlying frame-based vocoder, such as MELP, into a superframe structure. Parameters are extracted from the group of underlying frames and quantized into the superframe which allows the bit rate of the underlying coding to be reduced without increasing the distortion. The speech data coded in the superframe structure can then be directly synthesized to speech or may be transcoded to a format so that an underlying frame-based vocoder performs the synthesis. The superframe structure includes additional error detection and correction data to reduce the distortion caused by the communication of bit errors.

Abstract translation: 一种增强的低比特率参数语音编码器，其将来自基础帧的声码器（例如MELP）的多个帧分组成超帧结构。 从底层帧中提取参数并量化到超帧中，这允许在不增加失真的情况下减少底层编码的比特率。 然后，可以将在超帧结构中编码的语音数据直接合成为语音，或者将其转码为格式，使得基础的基于帧的声码器进行合成。 超帧结构包括附加的错误检测和校正数据，以减少由位错误的通信引起的失真。

公开(公告)号：WO9920021A3

公开(公告)日：1999-06-24

申请号：PCT/FI9800789

申请日：1998-10-09

Applicant: NOKIA TELECOMMUNICATIONS OY ,  HAEGGSTROEM JOHAN

Inventor： HAEGGSTROEM JOHAN

IPC: G10L19/16 ,  H04L12/56 ,  H04L12/64 ,  H04L12/66 ,  H04L29/06 ,  H04M7/00 ,  H04M7/12 ,  H04W28/18 ,  H04W80/00 ,  H04W88/18 ,  H04W92/02 ,  H04Q7/30

CPC classification number: H04W28/18 ,  G10L19/173 ,  H04L12/64 ,  H04L12/6418 ,  H04L29/06 ,  H04L29/06027 ,  H04L65/607 ,  H04L2012/6481 ,  H04M7/1235 ,  H04W80/00 ,  H04W88/04 ,  H04W88/16 ,  H04W88/181 ,  H04W92/02

Abstract: The invention concerns a system, which allows speech transmission between a mobile station and terminal equipment connected to a data network. The terminal equipment, preferably an Internet telephone, sends and receives data packets in accordance with the protocol of the data network in question and is provided with telephone characteristics. It has a speech coder in accordance with the mobile station system which synthesises the original speech from the speech parameters sent by the mobile station and contained in the data packets arriving from the data network and which correspondingly produces speech parameters for location in outgoing data packets. The speech parameters are conveyed as such between the mobile station and the terminal equipment without applying any additional coding to them. When the mobile telephone network is a packet switched network, it may be connected directly to the data network or, when the mobile telephone network is a circuit switched network, a gateway is used, which performs any necessary conversions, so that the call can be connected from one network to the other.

Abstract translation: 本发明涉及允许移动站与连接到数据网络的终端设备之间进行语音传输的系统。 终端设备，优选互联网电话，根据所讨论的数据网络的协议发送和接收数据分组，并提供电话特征。 它具有根据移动台系统的语音编码器，该语音编码器从移动台发送的语音参数合成原始语音，并包含在从数据网络到达的数据分组中，并且相应地产生出站数据分组中的位置的语音参数。 语音参数在移动台和终端设备之间传送，而不对其进行任何附加的编码。 当移动电话网络是分组交换网络时，其可以直接连接到数据网络，或者当移动电话网络是电路交换网络时，使用网关，其执行任何必要的转换，使得呼叫可以 从一个网络连接到另一个网络。

公开(公告)号：WO99020021A2

公开(公告)日：1999-04-22

申请号：PCT/FI1998/000789

申请日：1998-10-09

IPC: G10L19/16 ,  H04L12/56 ,  H04L12/64 ,  H04L12/66 ,  H04L29/06 ,  H04M7/00 ,  H04M7/12 ,  H04W28/18 ,  H04W80/00 ,  H04W88/18 ,  H04W92/02

CPC classification number: H04W28/18 ,  G10L19/173 ,  H04L12/64 ,  H04L12/6418 ,  H04L29/06 ,  H04L29/06027 ,  H04L65/607 ,  H04L2012/6481 ,  H04M7/1235 ,  H04W80/00 ,  H04W88/04 ,  H04W88/16 ,  H04W88/181 ,  H04W92/02

Abstract: The invention concerns a system, which allows speech transmission between a mobile station and terminal equipment connected to a data network. The terminal equipment, preferably an Internet telephone, sends and receives data packets in accordance with the protocol of the data network in question and is provided with telephone characteristics. It has a speech coder in accordance with the mobile station system which synthesises the original speech from the speech parameters sent by the mobile station and contained in the data packets arriving from the data network and which correspondingly produces speech parameters for location in outgoing data packets. The speech parameters are conveyed as such between the mobile station and the terminal equipment without applying any additional coding to them. When the mobile telephone network is a packet switched network, it may be connected directly to the data network or, when the mobile telephone network is a circuit switched network, a gateway is used, which performs any necessary conversions, so that the call can be connected from one network to the other.

Abstract translation: 本发明涉及一种用于在移动台和连接到数据网络的终端设备之间传输语音信号的系统。 终端设备，最好是因特网电话，根据所讨论的数据网络的协议发送和接收数据分组，并且具有电话特有的特性。 它包括合成从由产生语音设置在放置数据网络由移动站发送的和包含在数据分组的语音参数的原始声音的移动站系统下属的语音编码器 传出数据包。 语音参数在移动台和终端设备之间发送，而不分配附加的代码。 当移动电话网络是交换网络的分组时，它可以直接连接到数据网络或当移动电话网络是交换网络的电路，网关被用于执行所有必要的转换，以便 该呼叫可以从一个网络连接到另一个网络。

公开(公告)号：WO2017206432A1

公开(公告)日：2017-12-07

申请号：PCT/CN2016/104157

申请日：2016-10-31

Applicant: 华为技术有限公司

Inventor： 王宾 ,  夏丙寅 ,  刘泽新 ,  苗磊

IPC: G10L21/038

CPC classification number: H04M11/06 ,  G10L19/167 ,  G10L19/173 ,  G10L19/24 ,  G10L21/038 ,  H04M7/006 ,  H04N7/15

Abstract: 话音信号处理方法和相关装置及系统，话音信号处理方法包括：网络设备接收来自第一终端的第一话音编码信号（201），对第一话音编码信号进行话音解码处理以得到话音解码参数和第一话音解码信号（202）；利用话音解码参数进行虚拟频带扩展处理以得到与第一话音解码信号对应的扩频带话音解码信号（203）；将第一话音解码信号和扩频带话音解码信号组合后进行话音编码处理以得到第二话音编码信号（204）；向与第一终端建立了通话连接的第二终端发送第二话音编码信号（205），第一终端支持的最大频带带宽小于第二终端支持的最大频带带宽。有利于提升终端最大频带带宽支持能力非对称情况下的服务质量。

公开(公告)号：WO2017102560A1

公开(公告)日：2017-06-22

申请号：PCT/EP2016/080331

申请日：2016-12-08

Applicant: FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.

Inventor： NIEDERMEIER, Andreas ,  DISCH, Sascha

IPC: G10L19/16

CPC classification number: G10L19/173

Abstract: The invention refers to an apparatus for processing an encoded audio signal (100). The audio signal (100) comprises a sequence of access units (100'), each access unit comprising a core signal (101) with a first spectral width and parameters describing a spectrum above the first spectral width. The apparatus comprises: a demultiplexer (1) for generating, from an access unit (100') of the encoded audio signal (100), said core signal (101 ) and a set of said parameters (102), an upsampler (2) for upsampling said core signal (101 ) of said access unit (100') and outputting a first upsampled spectrum (103) and a timely consecutive second upsampled spectrum (103'), the first upsampled spectrum (103) and the second upsampled spectrum (103'), both, having a same content as the core signal (101 ) and having a second spectral width being greater than the first spectral width of the core spectrum (101), a parameter converter (3) for converting parameters of said set of parameters (102) of said access unit (100') to obtain converted parameters (104, 104'), and a spectral gap filling processor (4) for processing said first upsampled spectrum (103) and said second upsampled spectrum (103') using said converted parameters (104). The invention also refers to a corresponding method.

Abstract translation: 本发明涉及用于处理经编码的音频信号（100）的设备。 音频信号（100）包括访问单元（100'）的序列，每个访问单元包括具有第一频谱宽度的核心信号（101）和描述高于第一频谱宽度的频谱的参数。 该装置包括：解复用器（1），用于从编码音频信号（100）的存取单元（100'）产生所述核心信号（101）和一组所述参数（102）;上采样器（2） 用于对所述存取单元（100'）的所述核心信号（101）进行上采样并且输出第一上采样频谱（103）和及时连续第二上采样频谱（103'），第一上采样频谱（103）和第二上采样频谱 一个参数转换器（3），用于将所述组的参数（101,103'）与所述核心信号（101）的内容相同并具有大于所述核心频谱（101）的第一频谱宽度的第二频谱宽度; （103'）和所述第二上采样频谱（103'）的频谱间隙填充处理器（4），其中所述第一上采样频谱（103' ）使用所述转换后的参数（104）。 本发明还涉及相应的方法。

公开(公告)号：WO2017030705A1

公开(公告)日：2017-02-23

申请号：PCT/US2016/042648

申请日：2016-07-15

Applicant: QUALCOMM INCORPORATED

Inventor： ATTI, Venkatraman ,  CHEBIYYAM, Venkata Subrahmanyam Chandra Sekhar

IPC: G10L19/24 ,  G10L19/02

CPC classification number: G10L19/167 ,  G10L19/0208 ,  G10L19/173 ,  G10L19/24 ,  G10L25/06

Abstract: A method for generating a high-band target signal includes receiving, at an encoder, an input signal having a low-band portion and a high-band portion. The method also includes comparing a first autocorrelation value of the input signal to a second autocorrelation value of the input signal. The method further includes scaling the input signal by a scaling factor to generate a scaled input signal. The scaling factor is determined based on a result of the comparison. The method also includes generating a low-band signal based on the input signal and generating the high-band target signal based on the scaled input signal.

Abstract translation: 一种用于产生高频带目标信号的方法包括在编码器处接收具有低频带部分和高频带部分的输入信号。 该方法还包括将输入信号的第一自相关值与输入信号的第二自相关值进行比较。 该方法还包括按比例因子缩放输入信号以产生缩放的输入信号。 基于比较的结果确定比例因子。 该方法还包括基于输入信号生成低频带信号，并且基于经缩放的输入信号产生高频带目标信号。

公开(公告)号：WO2016204583A1

公开(公告)日：2016-12-22

申请号：PCT/KR2016/006497

申请日：2016-06-17

Applicant: 삼성전자 주식회사

Inventor： 김선민 ,  전상배

IPC: H04S3/00 ,  H04R5/04

CPC classification number: G10L19/173 ,  G10L19/008 ,  H04S3/008 ,  H04S2400/03

Abstract: 상기 기술적 과제를 해결하기 위한 본 발명의 일 실시예에 따른 오디오 신호를 처리하는 방법은, 인터널 채널 게인들(ICGs, Internal Channel Gain)이 선적용(pre-applied)된, 하나의 CPE(Channel Pair Element)에 대한 신호를 수신하는 단계; 재생 채널 구성이 스테레오가 아니라면, MPS212 파라미터들 및 포맷 컨버터에 정의된 MPS212 출력 채널들에 해당하는 렌더링 파라미터들에 기초하여 CPE에 대한 역 인터널 채널 게인들(inverse ICGs)을 획득하는 단계; 및 수신된 하나의 CPE에 대한 신호와 획득된 역 인터널 채널 게인들에 기초하여 출력 신호들을 생성하는 단계;를 더 포함한다.

Abstract translation: 根据用于实现技术目标的本发明的实施例的用于处理音频信号的方法包括以下步骤：接收内部信道增益（ICG）已被预先施加的信号，并且哪个用于一个信道对 元素（CPE）; 如果再现频道配置不是立体声，则根据与格式转换器中定义的MPS212输出声道和MPS212参数对应的渲染参数获得CPE的反内部通道增益（ICG）; 并根据一个CPE的接收信号和获得的反向内部信道增益产生输出信号。

公开(公告)号：WO2016062869A1

公开(公告)日：2016-04-28

申请号：PCT/EP2015/074623

申请日：2015-10-23

Applicant: DOLBY INTERNATIONAL AB

Inventor： KJOERLING, Kristofer ,  GROESCHEL, Alexander ,  PURNHAGEN, Heiko ,  HOERICH, Holger ,  KRAUSS, Kurt

IPC: G10L19/16 ,  G10L19/022

CPC classification number: G10L19/173 ,  G10L19/022 ,  G10L19/167 ,  H04N21/2335 ,  H04N21/23418

Abstract: An audio signal (X) is represented by a bitstream (B) segmented into frames. An audio processing system (500) comprises a buffer (510) and a decoding section (520). The buffer joins sets of audio data (D 1 ; D 2 ,..., D N ) carried by N respective frames (F 1 , F 2 ,..., F N ) into one decodable set of audio data (D) corresponding to a first frame rate and to a first number of samples of the audio signal per frame. The frames have a second frame rate corresponding to a second number of samples of the audio signal per frame. The first number of samples is N times the second number of samples. The decoding section decodes the decodable set of audio data into a segment of the audio signal by at least employing signal synthesis, based on the decodable set of audio data, with a stride corresponding to the first number of samples of the audio signal.

Abstract translation: 音频信号（X）由分割成帧的比特流（B）表示。 音频处理系统（500）包括缓冲器（510）和解码部分（520）。 缓冲器将由N个相应帧（F1，F2，...，FN）携带的音频数据集合（D1; D2，...，DN）连接成对应于第一帧速率的一个可解码的音频数据集合（D） 以及每帧的音频信号的第一数量的样本。 帧具有对应于每帧的音频信号的第二数量样本的第二帧速率。 样本的第一个数量是第二个样本数量的N倍。 解码部分通过至少采用基于可解码的音频数据集合的信号合成，将音频数据的可解码集合解码为音频信号的片段，其中步幅对应于音频信号的第一数量的样本。

公开(公告)号：WO2016030694A1

公开(公告)日：2016-03-03

申请号：PCT/GB2015/052501

申请日：2015-08-28

Applicant: LODE AUDIO LTD

Inventor： CUBITT, Dominic

IPC: H04L12/24 ,  H04L29/06 ,  G10L19/16 ,  H04J3/06

CPC classification number: H04J3/0664 ,  G10L19/173 ,  H04J3/0667 ,  H04L43/0858 ,  H04L65/602 ,  H04L65/608

Abstract: A system for transmitting low latency, synchronised audio that includes an audio source, a processor, a controller and a sink zone with a DAC. Particularly, the processor is capable of selectively resampling the audio source in order to output a data packet for transmission to the sink zone that has a maximised payload size while packet frequency remains a whole number.

Abstract translation: 用于传输具有DAC的低延迟同步音频的系统，其包括音频源，处理器，控制器和接收器区。 特别地，处理器能够选择性地重新采样音频源，以便输出用于发送到具有最大有效负载大小的信宿区域的数据分组，而分组频率保持整数。

公开(公告)号：WO2015003596A1

公开(公告)日：2015-01-15

申请号：PCT/CN2014/081763

申请日：2014-07-07

Applicant: TENCENT TECHNOLOGY (SHENZHEN) COMPANY LIMITED

Inventor： CHU, Pan ,  XU, Hao ,  WANG, Fei ,  HAN, Guiping ,  LIAO, Ling

IPC: G06F17/30

CPC classification number: G10L15/06 ,  G10L17/22 ,  G10L19/173 ,  G10L21/0364 ,  G10L21/04 ,  H04L51/12 ,  H04L51/28

Abstract: Systems,devices and methods are provided for configuring matching rules related to voice input commands. For example,a first mapping relation between one or more first original terms in a preset term database and one or more first identification terms is established; the first mapping relation is stored in a first mapping relation table; one or more first voice input commands are configured for the first identification terms or one or more first statements including the first identification terms; and a second mapping relation between the first identification terms or the first statements and the first voice input commands is stored into a second mapping relation table.

Abstract translation: 提供了系统，设备和方法来配置与语音输入命令相关的匹配规则。 例如，建立预设术语数据库中的一个或多个第一原始术语与一个或多个第一识别项之间的第一映射关系; 第一映射关系被存储在第一映射关系表中; 一个或多个第一语音输入命令被配置用于第一识别项或包括第一识别项的一个或多个第一语句; 并且第一识别项或第一语句和第一语音输入命令之间的第二映射关系被存储到第二映射关系表中。