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    • 3. 发明授权
    • Method for retransmitting a speech packet
    • 用于重传语音分组的方法
    • US07505764B2
    • 2009-03-17
    • US10695009
    • 2003-10-28
    • Lee Michael ProctorJames Patrick Ashley
    • Lee Michael ProctorJames Patrick Ashley
    • H04W24/00H04L12/66G10L21/00G10L19/12
    • H04L1/0014H04L1/1877H04W84/00
    • The invention provides a method of retransmitting a speech packet. In one embodiment, the method includes receiving (S702) at a transmitting device (140) a first negative acknowledgement (NACK) from a receiving device (120). The NAK indicates a corrupted first speech packet transmission. The transmitting device then retrieves (S706) a first speech packet associated with the first NACK and compresses (S714-S720) the speech packet to form a replacement speech packet. Next, a current segment of speech is encoded (S808) to form a current speech packet and current speech packet is combined with the replacements speech packet. The combined speech packet is then transmitted (S814) to the receiving device.
    • 本发明提供一种重传语音分组的方法。 在一个实施例中,该方法包括在发送设备(140)处接收来自接收设备(120)的第一否定确认(NACK)(S702)。 NAK表示损坏的第一语音分组传输。 然后,发送装置检索(S706)与第一NACK相关联的第一语音分组,并压缩(S714-S720)语音分组以形成替换语音分组。 接下来,对当前语音段进行编码(S808)以形成当前语音分组,并将当前语音分组与替换语音分组组合。 然后将组合的语音分组发送(S814)到接收设备。
    • 4. 发明授权
    • Method and system for controlling an encoding rate in a variable rate
communication system
    • 用于控制可变速率通信系统中的编码速率的方法和系统
    • US5898696A
    • 1999-04-27
    • US924602
    • 1997-09-05
    • Lee Michael ProctorJames Patrick Ashley
    • Lee Michael ProctorJames Patrick Ashley
    • H04B7/26H04B14/04H04J3/16H04L29/08H04Q11/04H04W28/22H04W88/18
    • H04W28/22H04B7/2659H04J3/1688H04W88/181
    • A method and system for controlling an encoding rate in a communication system utilizes feedforward rate information (48) and/or rate desirability information (50) sent from each of a plurality of variable rate vocoders (34) to a communication link output controller (24), such as a network arbitor. The communication link output controller (24) then sends a feedback control signal (32) to a selected variable rate vocoder to change the encoding rate of the selected variable rate vocoder to facilitate re-encoding of the speech packet when a bottleneck is detected. In another embodiment, the network arbitor (134) may additionally and independently modify speech packet data when it determines that a bottleneck may occur. The network arbitor (134) also communicates a packet modification control signal (132) for the variable rate vocoder that generated the dropped packet so that the corresponding variable rate vocoder can adjust its filter states to maintain convergence.
    • 用于控制通信系统中的编码速率的方法和系统利用从多个可变速率声码器(34)中的每一个发送到通信链路输出控制器(24)的前馈速率信息(48)和/或速率可取性信息(50) ),例如网络仲裁器。 然后,当检测到瓶颈时,通信链路输出控制器(24)然后向所选择的可变速率声码器发送反馈控制信号(32)以改变所选择的可变速率声码器的编码速率,以便于语音分组的重新编码。 在另一个实施例中,当网络仲裁器确定可能出现瓶颈时,网络仲裁器(134)可附加且独立地修改语音分组数据。 网络仲裁器(134)还传送用于生成丢弃的分组的可变速率声码器的分组修改控制信号(132),使得相应的可变速率声码器可以调整其过滤状态以保持收敛。
    • 7. 发明授权
    • Method and apparatus for suppressing acoustic background noise in a communication system by equaliztion of pre-and post-comb-filtered subband spectral energies
    • 通过前后梳状滤波的子带频谱能量的均衡来抑制通信系统中的声学背景噪声的方法和装置
    • US06366880B1
    • 2002-04-02
    • US09451074
    • 1999-11-30
    • James Patrick Ashley
    • James Patrick Ashley
    • G10L2102
    • G10L21/0208
    • A noise suppression system implemented in communication system provides an improved level of quality during severe signal-to-noise ratio (SNR) conditions. The noise suppression system, inter alia, incorporates a frequency domain comb-filtering (289) technique which supplements a traditional spectral noise suppression method. The invention includes a real cepstrum generator (285) for an input signal (285) G(k) to produce a likely voiced speech pitch lag component and converting a result to frequency domain to obtain a comb-filter function (290) C(k), applying input signal (291) G(k) to comb-filter function (290) C(k), and equalizing the energies of the corresponding pre and post filtered subbands, to produce a signal (293) G″(k) to be used for noise suppression. This prevents high frequency components from being unnecessarily attenuated, thereby reducing muffling effects of prior art comb-filters.
    • 在通信系统中实现的噪声抑制系统在严重的信噪比(SNR)条件下提供了改善的质量水平。 噪声抑制系统尤其包含频域梳状滤波(289)技术,其补充传统的频谱噪声抑制方法。 本发明包括用于输出信号(285)G(k)的真实倒谱发生器(285),以产生可能的有声语音音调滞后分量并将结果转换到频域以获得梳状滤波器函数(290)C(k ),将输入信号(291)G(k)应用于梳状滤波器函数(290)C(k),并且对相应的前和后滤波后的子带的能量进行均衡,以产生信号(293)G“ )用于噪声抑制。 这防止高频分量被不必要地衰减,从而减少现有技术梳状滤波器的消音效果。
    • 8. 发明授权
    • Method and apparatus for coding and decoding speech
    • 用于语音编码和解码的方法和装置
    • US06415252B1
    • 2002-07-02
    • US09086396
    • 1998-05-28
    • Weimin PengJames Patrick Ashley
    • Weimin PengJames Patrick Ashley
    • G10L1106
    • G10L19/04G10L25/93G10L2019/0004
    • Bits are allocated to short-term repetition information for unvoiced input signals. Stated differently, more bits are allocated for pitch information during unvoiced input speech than in the prior art. The improved method and apparatus in an encoder (300) and decoder (700) result in improved consistency of amplitude pulses compared to prior art methods which indicates improved stability due to increased search resolution. Also, the improved method and apparatus result in higher energy compared to prior art methods which indicates that the synthesized waveform matches the target waveform more closely, resulting in a higher fixed codebook (FCB) gain.
    • 位被分配给无声输入信号的短期重复信息。 换句话说,在清音输入语音中比现有技术更多的比特被分配用于音调信息。 与现有技术方法相比,编码器(300)和解码器(700)中改进的方法和装置导致振幅脉冲的一致性提高,这表明由于增加的搜索分辨率而提高了稳定性。 此外,与现有技术方法相比,改进的方法和装置导致更高的能量,这表明合成波形更加紧密地匹配目标波形,导致更高的固定码本(FCB)增益。