ACOUSTIC SHOCK PROTECTION DEVICE AND METHOD THEREOF
    71.
    发明申请
    ACOUSTIC SHOCK PROTECTION DEVICE AND METHOD THEREOF 有权
    震动保护装置及其方法

    公开(公告)号:US20130030799A1

    公开(公告)日:2013-01-31

    申请号:US13557230

    申请日:2012-07-25

    IPC分类号: G10L19/04

    摘要: An acoustic shock protection device includes a prediction gain estimator and an audio compressor. The prediction gain estimator is configured to analyze a plurality of linear prediction coefficients of an audio signal and determine a category of the audio signal. The audio compressor is coupled to the prediction gain estimator, and the audio compressor is configured to adjust a signal level of the audio signal according to the category of the audio signal.

    摘要翻译: 声学防震装置包括预测增益估计器和音频压缩器。 预测增益估计器被配置为分析音频信号的多个线性预测系数并确定音频信号的类别。 音频压缩器耦合到预测增益估计器,并且音频压缩器被配置为根据音频信号的类别来调整音频信号的信号电平。

    Speech enhancement system
    72.
    发明授权
    Speech enhancement system 有权
    语音增强系统

    公开(公告)号:US08326614B2

    公开(公告)日:2012-12-04

    申请号:US12471072

    申请日:2009-05-22

    IPC分类号: G10L19/04

    摘要: A speech enhancement system improves speech conversion within an encoder and decoder. The system includes a first device that converts sound waves into operational signals. A second device selects a template that represents an expected signal model. The selected template models speech characteristics of the operational signals through a speech codebook that is further accessed in a communication channel.

    摘要翻译: 语音增强系统改进编码器和解码器内的语音转换。 该系统包括将声波转换成操作信号的第一装置。 第二个设备选择一个表示预期信号模型的模板。 所选择的模板通过在通信信道中进一步访问的语音码本来模拟操作信号的语音特征。

    METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL
    74.
    发明申请
    METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL 有权
    用于处理音频信号的方法和装置

    公开(公告)号:US20120239408A1

    公开(公告)日:2012-09-20

    申请号:US13391992

    申请日:2010-09-17

    IPC分类号: G10L19/04

    摘要: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, coding identification information indicating whether to apply a first coding scheme or a second coding scheme to a current frame; when the coding identification information indicates that the second coding scheme is applied to the current frame, receiving window type information indicating a particular window for the current frame, from among a plurality of windows; identifying that a current window is a long stop window based on the window type information, wherein the long stop window is followed by only long window of a following frame, wherein the long stop window includes a gentle long stop window and a steep long stop window; and, when the first coding scheme is applied to a previous frame, applying the gentle long stop window to the current frame, wherein: the gentle long stop window comprise an ascending line with first slope, the steep long stop window comprise an ascending line with second slope, and, the first slope is gentler than the second slope.

    摘要翻译: 公开了一种处理音频信号的方法。 本发明包括由音频处理装置接收指示是否对当前帧应用第一编码方案或第二编码方案的编码标识信息; 当编码识别信息指示第二编码方案被应用于当前帧时,从多个窗口中接收指示当前帧的特定窗口的窗口类型信息; 基于所述窗口类型信息识别当前窗口是长时间停止窗口,其中所述长停止窗口后面仅是后续框架的长窗口,其中所述长停止窗口包括平缓的长停止窗口和陡峭的长停止窗口 ; 并且当将第一编码方案应用于先前帧时,将柔和的长时间窗口应用于当前帧,其中:平缓的长停止窗口包括具有第一斜率的上行线,陡峭的长停止窗口包括具有 第二斜坡,第一斜坡比第二斜坡慢。

    CONTEXT-BASED ARITHMETIC ENCODING APPARATUS AND METHOD AND CONTEXT-BASED ARITHMETIC DECODING APPARATUS AND METHOD
    75.
    发明申请
    CONTEXT-BASED ARITHMETIC ENCODING APPARATUS AND METHOD AND CONTEXT-BASED ARITHMETIC DECODING APPARATUS AND METHOD 有权
    基于语境的算术编码设备和方法和基于语境的算术解码设备和方法

    公开(公告)号:US20120221325A1

    公开(公告)日:2012-08-30

    申请号:US13464529

    申请日:2012-05-04

    IPC分类号: G10L19/04

    摘要: Disclosed are a context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code.

    摘要翻译: 公开了一种基于上下文的算术编码装置和方法以及基于上下文的算术解码装置和方法。 基于上下文的算术解码装置可以确定要解码的当前N元组的上下文,确定与当前N元组的MSB符号相对应的最高有效位(MSB)上下文,并使用上下文确定概率模型 的N元组和MSB上下文。 随后,基于上下文的算术解码装置可以基于所确定的概率模型对MSB执行解码,并且基于从解码处理导出的LSB的位深度对最低有效位(LSB)执行解码 一个转义码。

    Method and apparatus for decoding an audio signal using a rendering parameter
    76.
    发明授权
    Method and apparatus for decoding an audio signal using a rendering parameter 有权
    使用渲染参数对音频信号进行解码的方法和装置

    公开(公告)号:US08239209B2

    公开(公告)日:2012-08-07

    申请号:US12161331

    申请日:2007-01-19

    摘要: An apparatus for decoding a signal and method thereof are disclosed, by which the audio signal can be controlled in a manner of changing/giving spatial characteristics (e.g., listener's virtual position, virtual position of a specific source) of the audio signal. The present invention includes receiving an object parameter including level information corresponding to at least one object signal, converting the level information corresponding to the object signal to the level information corresponding to an output channel by applying a control parameter to the object parameter, and generating a rendering parameter including the level information corresponding to the output channel to control an object downmix signal resulting from downmixing the object signal.

    摘要翻译: 公开了用于对信号进行解码的装置及其方法,通过该装置可以以改变/给出音频信号的空间特性(例如,听众的虚拟位置,特定源的虚拟位置)的方式来控制音频信号。 本发明包括:接收包括与至少一个对象信号相对应的级别信息的对象参数,通过对对象参数应用控制参数将与对象信号对应的级别信息转换为对应于输出通道的级别信息,以及生成 渲染参数包括对应于输出通道的电平信息,以控制由对象信号降混得到的对象下混信号。

    SPEECH ENCODING/DECODING DEVICE
    77.
    发明申请
    SPEECH ENCODING/DECODING DEVICE 有权
    语音编码/解码设备

    公开(公告)号:US20120010879A1

    公开(公告)日:2012-01-12

    申请号:US13243015

    申请日:2011-09-23

    IPC分类号: G10L19/04

    摘要: A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is transformed. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain represented by SBR.

    摘要翻译: 通过使用协方差方法或自相关方法通过在频率方向上进行线性预测分析来获得在频域中表示的信号的线性预测系数。 在调整所获得的线性预测系数的滤波强度之后,可以通过使用调整后的系数在频率方向对信号进行滤波,从而变换信号的时间包络。 这减少了前回波和后回波的发生,并提高了解码信号的主观质量,而不会显着增加由SBR表示的频域中的频带扩展技术中的比特率。

    Low complexity no delay reconstruction of missing packets for LPC decoder
    78.
    发明授权
    Low complexity no delay reconstruction of missing packets for LPC decoder 有权
    低复杂度无延迟重建LPC解码器的丢包

    公开(公告)号:US07991612B2

    公开(公告)日:2011-08-02

    申请号:US11927512

    申请日:2007-10-29

    IPC分类号: G10L19/04

    摘要: Lost frame reconstruction is described. A previous good or reconstructed frame may be analyzed to determine a category for the lost frame. A percentage Pi may be associated with the determined category of the lost frame. A top Pi percent magnitude samples may be zeroed out in an excitation of the previous good or reconstructed frame to produce a reconstruction excitation. The reconstruction excitation may be applied to one or more linear prediction coefficients for the previous good or reconstructed frame to generate a reconstructed frame.

    摘要翻译: 描述丢失帧重建。 可以分析先前的好的或重构的帧以确定丢失帧的类别。 百分比Pi可以与确定的丢失帧的类别相关联。 在先前的好的或重建的帧的激励中可以将顶部的Pi百分比幅度样本归零,以产生重建激励。 重构激励可以应用于先前的好的或重构的帧的一个或多个线性预测系数,以产生重建的帧。

    METHOD, APPARATUS, PROGRAM AND RECORDING MEDIUM FOR LONG-TERM PREDICTION CODING AND LONG-TERM PREDICTION DECODING
    79.
    发明申请
    METHOD, APPARATUS, PROGRAM AND RECORDING MEDIUM FOR LONG-TERM PREDICTION CODING AND LONG-TERM PREDICTION DECODING 有权
    用于长期预测编码和长期预测解码的方法,装置,程序和记录介质

    公开(公告)号:US20110166854A1

    公开(公告)日:2011-07-07

    申请号:US13049442

    申请日:2011-03-16

    IPC分类号: G10L19/04

    摘要: A method and apparatus multiplies a past sample a time lag τ older than a current sample by a quantized multiplier ρ′ on a frame by frame basis, subtracts the multiplication result from the current sample, codes the subtraction result, and codes the time lag using a fixed-length coder if the multiplier ρ′ is smaller than 0.2 or if information about the previous frame is unavailable, or codes the time lag using a variable-length coder if ρ′ is not smaller than 0.2. A multiplier ρ is coded by a multiplier coder and the multiplier ρ′ obtained by decoding the multiplier ρ is outputted. The process is performed for each frame.

    摘要翻译: 一种方法和装置将一个比当前样本更早的时间滞后τ乘以一个逐帧的量化乘法器,从当前样本中减去相乘结果,对相减结果进行编码,并对其进行编码 如果乘数&rgr''小于0.2,或者如果关于前一帧的信息不可用,则使用固定长度编码器,如果&rgr;'不小于0.2,则使用可变长度编码器对时间延迟进行编码。 乘法器 由乘法器编码器和乘法器&rgr;'通过对乘法器&rgr进行解码获得; 被输出。 对每个帧执行该过程。

    Method and System for Frequency Domain Postfiltering of Encoded Audio Data in a Decoder
    80.
    发明申请
    Method and System for Frequency Domain Postfiltering of Encoded Audio Data in a Decoder 审中-公开
    解码器中编码音频数据的频域后置滤波方法与系统

    公开(公告)号:US20110125507A1

    公开(公告)日:2011-05-26

    申请号:US13054518

    申请日:2009-07-14

    申请人: Rongshan Yu

    发明人: Rongshan Yu

    IPC分类号: G10L19/00 G10L19/04

    摘要: A decoder configured to generate decoded audio data (e.g., decoded speech data) and including a postfilter coupled and configured to filter encoded audio data in the frequency domain, methods for frequency domain postfiltering of encoded audio data in a decoder, and methods for decoding encoded audio data in a decoder including by postfiltering encoded audio data in the frequency domain in the decoder. In some embodiments, the decoder is configured to decode input encoded audio without performing any time-to-frequency domain transform on encoded audio data to prepare data for postfiltering. Typically, the postfiltering improves the quality of the decoded audio signal by attenuating spectral valley regions thereof to remove excess quantization noise present in the encoded input audio while preserving formants of the decoded audio signal to avoid introducing unnecessary distortion.

    摘要翻译: 解码器,被配置为生成解码的音频数据(例如,解码的语音数据),并且包括被耦合并被配置为对频域中的编码的音频数据进行滤波的后置滤波器,用于解码器中的编码音频数据的频域后置滤波的方法, 解码器中的音频数据包括通过对解码器中的频域中的编码音频数据进行后置滤波。 在一些实施例中,解码器被配置为对输入的编码音频进行解码,而不对经编码的音频数据执行任何时间 - 频域变换以准备用于后置滤波的数据。 通常,后置滤波通过衰减其解码音频信号的质量,从而消除编码的输入音频中存在的多余的量化噪声,同时保留解码音频信号的共振峰,以避免引入不必要的失真。