High frequency and low frequency audio signal encoding and decoding system
    1.
    发明授权
    High frequency and low frequency audio signal encoding and decoding system 有权
    高频和低频音频信号编解码系统

    公开(公告)号:US06772114B1

    公开(公告)日:2004-08-03

    申请号:US09710916

    申请日:2000-11-13

    IPC分类号: G10L1900

    CPC分类号: G10L19/0208

    摘要: In an audio transmission system, an input signal is split up into two spectral portions in a transmitter. These spectral portions are coded by their own respective coder. The low-frequency signal portion is coded by a regular narrow-band coder and the high frequency portion is coded using a coder that outputs LPC codes and signal amplitude codes. In the receiver, the low frequency signal portion is reconstructed by a narrow-band decoder and the high frequency portion is reconstructed by applying a high pass filter to a white noise signal and applying an LPC filter that is controlled by the LPC codes to this filtered white noise signal and adjusting the signal amplitude with an amplifier that is controlled using the amplitude codes of the transmitter. The reconstructed low frequency signal and the reconstructed high frequency signal are then combined to yield a reconstructed output signal containing both frequency ranges.

    摘要翻译: 在音频传输系统中,输入信号在发射机中被分成两个频谱部分。 这些频谱部分由其各自的编码器编码。 低频信号部分由常规窄带编码器编码,并且使用输出LPC码和信号幅度码的编码器对高频部分进行编码。 在接收机中,低频信号部分由窄带解码器重构,高频部分通过对白噪声信号施加高通滤波器并将由LPC码控制的LPC滤波器应用于该滤波器 白噪声信号,并用放大器调节信号幅度,该放大器使用发射机的幅度码进行控制。 然后将重构的低频信号和重建的高频信号组合以产生包含两个频率范围的重构输出信号。

    Wideband extension of telephone speech for higher perceptual quality
    2.
    发明授权
    Wideband extension of telephone speech for higher perceptual quality 失效
    宽带扩展电话语音,提高感知质量

    公开(公告)号:US07346499B2

    公开(公告)日:2008-03-18

    申请号:US10169497

    申请日:2001-11-09

    IPC分类号: G10L21/00

    CPC分类号: G10L21/0364 G10L25/24

    摘要: Wideband extension of telephone speech for higher perceptual quality. A method for extending the frequency range of a speech signal using wideband extension method with an inverse filter and a synthesis filter where both filters receive LPC coefficients from an LPC estimator. The wideband LPC coefficients are obtained from wideband LSFs. The wideband LSFs are obtained by appending highband LSFs, created by applying a matrix to narrowband LSFs, and lowband LSFs, created by dividing the narrowband LSFs by two. The matrix used to create the highband LSFs is selected from a predetermined list of matrices. The selection is based on either wideband or narrowband reflection coefficients extracted from the narrowband speech signal.

    摘要翻译: 宽带扩展电话语音,提高感知质量。 一种使用宽带扩展方法用反向滤波器和合成滤波器扩展语音信号的频率范围的方法,其中两个滤波器从LPC估计器接收LPC系数。 从宽带LSF获得宽带LPC系数。 通过将通过将窄带LSF应用矩阵创建的高频LSF和通过将窄带LSF除以2而创建的低频LSF来获得宽带LSF。 用于创建高频LSF的矩阵从预定的矩阵列表中选择。 该选择基于从窄带语音信号提取的宽带或窄带反射系数。

    Wideband signal transmission system
    3.
    发明授权
    Wideband signal transmission system 有权
    宽带信号传输系统

    公开(公告)号:US07174135B2

    公开(公告)日:2007-02-06

    申请号:US10480660

    申请日:2002-06-20

    IPC分类号: H04B1/00 H04B7/00 G10L19/00

    CPC分类号: G10L21/038

    摘要: Described is a transmission system (10) comprising a transmitter (12) for transmitting a narrowband audio signal to a receiver (14) via a transmission channel (16). The receiver (14) comprises a frequency domain bandwidth extender (18) for extending a bandwidth of the received narrowband audio signal by complementing the received narrowband audio signal with a highband extension thereof. The bandwidth extender (18) comprises an amplitude extender (24) for extending the bandwidth of an amplitude spectrum of the received narrowband audio signal by mapping narrowband amplitudes onto highband amplitudes. The bandwidth extender (18) further comprises a phase extender (26) for extending the bandwidth of a phase spectrum of the received narrowband signal and a combiner (28) for combining the extended amplitude spectrum and the extended phase spectrum into a bandwidth extended audio signal. The transmission system (10) is characterized in that the amplitude extender (24) comprises an amplitude mapper (42) and first and second frequency scale transformers (40,44). The first frequency scale transformer. (40) is arranged for transforming a linear frequency scale of the amplitude spectrum into a logarithmic frequency scale, e.g. the Bark scale. The amplitude mapper (42) is arranged for mapping according to the logarithmic frequency scale the narrowband amplitudes onto the highband amplitudes. The second frequency scale transformer (44) is arranged for transforming the logarithmic frequency scale of the extended amplitude spectrum into the linear frequency scale.

    摘要翻译: 描述了一种传输系统(10),包括用于经由传输信道(16)将窄带音频信号发送到接收机(14)的发射机(12)。 接收器(14)包括频域带宽扩展器(18),用于通过用接收的窄带音频信号的高频扩展来补充接收到的窄带音频信号来扩展接收的窄带音频信号的带宽。 带宽扩展器(18)包括幅度扩展器(24),用于通过将窄带幅度映射到高频带幅度上来扩展接收的窄带音频信号的幅度谱带宽。 带宽扩展器(18)还包括扩展器(26),用于扩展所接收的窄带信号的相位频带的带宽;以及组合器(28),用于将扩展幅度频谱和扩展相位频谱组合成带宽扩展音频信号 。 传输系统(10)的特征在于,幅度扩展器(24)包括幅度映射器(42)和第一和第二频率比例变换器(40,44)。 第一台变频器。 (40)被布置用于将幅度谱的线性频率变换变换成对数频率标度,例如。 吠声鳞片。 幅度映射器(42)被布置用于根据对数频率标度将窄带幅度映射到高频带幅度上。 第二频率比例变换器(44)被设置为将扩展振幅频谱的对数频谱变换为线性频率标度。

    Audio coding via creation of sinusoidal tracks and phase determination
    4.
    发明授权
    Audio coding via creation of sinusoidal tracks and phase determination 有权
    通过创建正弦曲线进行音频编码和相位确定

    公开(公告)号:US07664633B2

    公开(公告)日:2010-02-16

    申请号:US10536228

    申请日:2003-11-06

    CPC分类号: G10L19/093

    摘要: Coding of an audio signal represented by a respective set of sampled signal values for each of a plurality of sequential segments is disclosed. The sampled signal values are analyzed (40) to determine one or more sinusoidal components for each of the plurality of sequential segments. The sinusoidal components are linked (42) across a plurality of sequential segments to provide sinusoidal tracks. For each sinusoidal track, a phase comprising a generally monotonically changing value is determined and an encoded audio stream including sinusoidal codes (r) representing said phase is generated (46).

    摘要翻译: 公开了由多个连续段中的每一个的相应的一组采样信号值表示的音频信号的编码。 分析采样信号值(40)以确定多个连续段中的每一个的一个或多个正弦分量。 正弦分量连接(42)跨越多个顺序段以提供正弦曲线。 对于每个正弦曲线,确定包括通常单调变化的值的相位,并且生成包括表示所述相位的正弦码(r)的编码音频流(46)。

    Time-scale modification of signals
    5.
    发明授权
    Time-scale modification of signals 失效
    时间尺度修改信号

    公开(公告)号:US07412379B2

    公开(公告)日:2008-08-12

    申请号:US10114505

    申请日:2002-04-02

    IPC分类号: G10L11/06

    CPC分类号: G10L21/04 G10L25/93

    摘要: Techniques utilising Time Scale Modification (TSM) of signals are described. The signal is analysed and divided into frames of similar signal types. Techniques specific to the signal type are then applied to the frames thereby optimising the modification process. The method of the present invention enables TSM of different audio signal parts to be realized using different methods, and a system for effecting said method is also described.

    摘要翻译: 描述了使用信号的时间尺度修正(TSM)的技术。 信号被分析并分成类似信号类型的帧。 然后将特定于信号类型的技术应用于帧,从而优化修改过程。 本发明的方法能够使用不同的方法实现不同的音频信号部分的TSM,并且还描述了用于实现所述方法的系统。

    Communication system having bad frame indicator means for resynchronization purposes
    6.
    发明授权
    Communication system having bad frame indicator means for resynchronization purposes 失效
    具有不良帧指示符的通信系统用于重新同步化

    公开(公告)号:US06941150B2

    公开(公告)日:2005-09-06

    申请号:US09989256

    申请日:2001-11-20

    摘要: A communication system (1) comprises a transmitter (2), a receiver (3), and an up/down link communication channel (4, 6) arranged for data communication from the transmitter (2) through the up link communication channel (4) to the receiver (3). The communication system (1) is further arranged to feedback data from the receiver (3) through the down link communication channel (6) to the transmitter (2). The receiver (3) comprises a bad frame indicator (5) for providing a bad frame indication (BFI) upon receipt of a corrupted frame, which is present in synchronized data communicated over the up link communication channel (4); and the transmitter (2) comprises resynchronization means (7) coupled to the down link communication channel (6) for receiving BFI related data and in response thereto recommencing data communication over the up link communication channel (4), in accordance with a resynchronization procedure, which starts from a predetermined state. A fast acting feedback resynchronization procedure for a GSM speech system is presented which prevents substantial error propagation from occurring at the receiver end.

    摘要翻译: 通信系统(1)包括发射机(2),接收机(3)和布置用于通过上行链路通信信道(4)从发射机(2)进行数据通信的上/下链路通信信道 )到接收器(3)。 通信系统(1)还被布置为通过下行链路通信信道(6)将数据从接收机(3)反馈到发射机(2)。 接收器(3)包括用于在接收到通过上行链路通信信道(4)传送的同步数据中存在的损坏的帧时提供坏帧指示(BFI)的坏帧指示符(5)。 并且所述发射机(2)包括耦合到所述下行链路通信信道(6)的再同步装置(7),用于接收BFI相关数据,并且响应于所述重新同步装置根据重新同步过程在所述上行链路通信信道(4)上重新开始数据通信 ,其从预定状态开始。 提出了一种用于GSM语音系统的快速反馈重新同步过程,其防止在接收机端发生大量错误传播。

    Sound synthesis
    7.
    发明授权
    Sound synthesis 有权
    声音综合

    公开(公告)号:US07781665B2

    公开(公告)日:2010-08-24

    申请号:US11908321

    申请日:2006-02-01

    IPC分类号: G10H7/00

    摘要: A device (1) is arranged for synthesizing sound represented by sets of parameters, each set comprising noise parameters (NP) representing noise components of the sound and optionally also other parameters representing other components, such as transients and sinusoids. Each set of parameters may correspond with a sound channel, such as a MIDI voice. In order to reduce the computational load, the device comprises a selection unit (2) for selecting a limited number of sets from the total number of sets on the basis of a perceptual relevance value, such as the amplitude or energy. The device further comprises a synthesizing unit (3) for synthesizing the noise components using the noise parameters of the selected sets only.

    摘要翻译: 设备(1)被布置为用于合成由参数集合表示的声音,每组包括表示声音的噪声分量的噪声参数(NP),以及可选地还包括表示其他分量的其它参数,例如瞬变和正弦波。 每组参数可以对应于诸如MIDI声音之类的声音通道。 为了减少计算负荷,该装置包括一个选择单元(2),用于根据诸如振幅或能量之类的感知相关性值从总数量集中选择有限数量的集合。 该装置还包括用于仅使用所选集合的噪声参数合成噪声分量的合成单元(3)。

    Coding Reverberant Sound Signals
    8.
    发明申请
    Coding Reverberant Sound Signals 审中-公开
    编码混响声音信号

    公开(公告)号:US20080281602A1

    公开(公告)日:2008-11-13

    申请号:US11569778

    申请日:2005-06-03

    IPC分类号: G10L19/00

    CPC分类号: G10L19/00

    摘要: The invention relates to an audio encoder and decoder and methods for audio encoding and decoding. In the encoder an audio signal is split into an anechoic signal part and information regarding a reverberant field associated with the audio signal, preferably by a representation using only few parameters such as reverberation time and reverberation amplitude. The anechoic signal is then encoded using an audio codec. At the decoder the anechoic signal part is restored using the audio codec, and the restored anechoic signal part is transformed into the substantially original audio signal by applying reverberance according to the information regarding the reverberant field, preferably by convolution with a room impulse response generated on the basis of the reverberant field information. According to the invention the audio codec involved needs only be capable of encoding anechoic audio signals, thus solving the problem of parametric audio codecs providing poor performance on reverberant audio signals.

    摘要翻译: 本发明涉及音频编码器和解码器以及用于音频编码和解码的方法。 在编码器中,音频信号被分成消音信号部分和关于与音频信号相关联的混响场的信息,优选地仅通过使用诸如混响时间和混响幅度之类的少量参数的表示。 然后使用音频编解码器对消声信号进行编码。 在解码器处,使用音频编解码器恢复消声信号部分,并且通过根据关于混响场的信息应用混响,将恢复的消声信号部分变换为基本上原始的音频信号,优选地通过与在 混响场地信息的基础。 根据本发明,所涉及的音频编解码器仅需要能够对无回音音频信号进行编码,从而解决在混响音频信号上提供差的性能的参数音频编解码器的问题。

    Parametric Multi-Channel Decoding
    9.
    发明申请
    Parametric Multi-Channel Decoding 审中-公开
    参数多通道解码

    公开(公告)号:US20080212784A1

    公开(公告)日:2008-09-04

    申请号:US11994458

    申请日:2006-07-03

    IPC分类号: H04R5/00

    摘要: A sound decoding device (1) is arranged for decoding sound represented by sets of parameters, each set comprising sinusoidal parameters (SP) representing sinusoidal components of the sound and further parameters (NP, TP) representing further components of the sound, such as noise and/or transients. The device comprises a separate sinusoids generator unit (17, 18) for each output channel (L, R), while the further component generator units (20; 21) are shared between the channels.

    摘要翻译: 声音解码装置(1)被布置成用于解码由参数集合表示的声音,每个组包括表示声音的正弦分量的正弦参数(SP)和表示声音的其它分量的其它参数(NP,TP),例如噪声 和/或瞬变。 该装置包括用于每个输出通道(L,R)的单独的正弦波发生器单元(17,18),而另外的组件发生器单元(20; 21)在通道之间共享。