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    • 1. 发明申请
    • Robust real-time speech codec
    • 强大的实时语音编解码器
    • US20050228651A1
    • 2005-10-13
    • US10816466
    • 2004-03-31
    • Tian WangHosam KhalilKazuhito KoishidaWei-Ge ChenMu Han
    • Tian WangHosam KhalilKazuhito KoishidaWei-Ge ChenMu Han
    • G10L11/06G10L19/08
    • G10L19/08G10L19/005G10L19/22
    • Various strategies for rate/quality control and loss resiliency in an audio codec are described. The various strategies can be used in combination or independently. For example, a real-time speech codec uses intra frame coding/decoding, adaptive multi-mode forward error correction [“FEC”], and rate/quality control techniques. Intra frames help a decoder recover quickly from packet losses, while compression efficiency is still emphasized with predicted frames. Various strategies for inserting intra frames and signaling intra/predicted frames are described. With the adaptive multi-mode FEC, an encoder adaptively selects between multiple modes to efficiently and quickly provide a level of FEC that takes into account the bandwidth currently available for FEC. The FEC information itself may be predictively encoded and decoded relative to primary encoded information. Various rate/quality and FEC control strategies allow additional adaptation to available bandwidth and network conditions.
    • 描述了音频编解码器中的速率/质量控制和丢失弹性的各种策略。 各种策略可以组合使用或独立使用。 例如,实时语音编解码器使用帧内编码/解码,自适应多模式前向纠错[“FEC”]和速率/质量控制技术。 帧内帧帮助解码器从分组丢失中快速恢复,而预测帧仍然强调压缩效率。 描述了用于插入帧内和信令帧内/预测帧的各种策略。 利用自适应多模式FEC,编码器在多种模式之间自适应地选择以有效且快速地提供考虑到当前可用于FEC的带宽的FEC级别。 FEC信息本身可以相对于主编码信息进行预测编码和解码。 各种速率/质量和FEC控制策略允许对可用带宽和网络条件进行额外的调整。
    • 9. 发明申请
    • Gain constrained noise suppression
    • 增加约束噪声抑制
    • US20050278172A1
    • 2005-12-15
    • US10869467
    • 2004-06-15
    • Kazuhito KoishidaFeng ZhugeHosam KhalilTian WangWei-ge Chen
    • Kazuhito KoishidaFeng ZhugeHosam KhalilTian WangWei-ge Chen
    • G10L15/20G10L21/02
    • G10L21/0208G10L21/0232
    • A gain-constrained noise suppression for speech more precisely estimates noise, including during speech, to reduce musical noise artifacts introduced from noise suppression. The noise suppression operates by applying a spectral gain G(m, k) to each short-time spectrum value S(m, k) of a speech signal, where m is the frame number and k is the spectrum index. The spectrum values are grouped into frequency bins, and a noise characteristic estimated for each bin classified as a “noise bin.” An energy parameter is smoothed in both the time domain and the frequency domain to improve noise estimation per bin. The gain factors G(m, k) are calculated based on the current signal spectrum and the noise estimation, then smoothed before being applied to the signal spectral values S(m, k). First, a noisy factor is computed based on a ratio of the number of noise bins to the total number of bins for the current frame, where a zero-valued noisy factor means only using constant gain for all the spectrum values and noisy factor of one means no smoothing at all. Then, this noisy factor is used to alter the gain factors, such as by cutting off the high frequency components of the gain factors in the frequency domain.
    • 用于语音的增益约束噪声抑制更精确地估计包括在语音期间的噪声,以减少从噪声抑制引入的音乐噪声伪像。 通过对语音信号的每个短时间频谱值S(m,k)应用频谱增益G(m,k)来进行噪声抑制,其中m是帧号,k是频谱索引。 频谱值被分组成频率仓,并且对于被分类为“噪声仓”的每个仓估计的噪声特性。 能量参数在时域和频域均被平滑,以改善每个bin的噪声估计。 基于当前信号频谱和噪声估计来计算增益因子G(m,k),然后在施加到信号频谱值S(m,k)之前进行平滑处理。 首先,基于噪声箱数与当前帧的总数的比率来计算噪声因子,其中零值噪声因子意味着仅对所有频谱值使用恒定增益并且噪声因子为1 意味着没有平滑。 然后,这种噪声因子用于改变增益因子,例如通过切断频域中增益因子的高频分量。
    • 10. 发明申请
    • AUDIO ENCODING AND DECODING WITH INTRA FRAMES AND ADAPTIVE FORWARD ERROR CORRECTION
    • 音频编码和解码与内部框架和自适应前向错误校正
    • US20100125455A1
    • 2010-05-20
    • US12692417
    • 2010-01-22
    • Tian WangHosam A. KhalilKazuhito KoishidaWei-Ge ChenMu Han
    • Tian WangHosam A. KhalilKazuhito KoishidaWei-Ge ChenMu Han
    • G10L19/08
    • G10L19/08G10L19/005G10L19/22
    • Various strategies for rate/quality control and loss resiliency in an audio codec are described. The various strategies can be used in combination or independently. For example, a real-time speech codec uses intra frame coding/decoding, adaptive multi-mode forward error correction [“FEC”], and rate/quality control techniques. Intra frames help a decoder recover quickly from packet losses, while compression efficiency is still emphasized with predicted frames. Various strategies for inserting intra frames and signaling intra/predicted frames are described. With the adaptive multi-mode FEC, an encoder adaptively selects between multiple modes to efficiently and quickly provide a level of FEC that takes into account the bandwidth currently available for FEC. The FEC information itself may be predictively encoded and decoded relative to primary encoded information. Various rate/quality and FEC control strategies allow additional adaptation to available bandwidth and network conditions.
    • 描述了音频编解码器中的速率/质量控制和丢失弹性的各种策略。 各种策略可以组合使用或独立使用。 例如,实时语音编解码器使用帧内编码/解码,自适应多模式前向纠错[“FEC”]和速率/质量控制技术。 帧内帧帮助解码器从分组丢失中快速恢复,而预测帧仍然强调压缩效率。 描述了用于插入帧内和信令帧内/预测帧的各种策略。 利用自适应多模式FEC,编码器在多种模式之间自适应地选择以有效且快速地提供考虑到当前可用于FEC的带宽的FEC级别。 FEC信息本身可以相对于主编码信息进行预测编码和解码。 各种速率/质量和FEC控制策略允许对可用带宽和网络条件进行额外的调整。