公开(公告)号：US20060241942A1

公开(公告)日：2006-10-26

申请号：US11475302

申请日：2006-06-26

申请人： Wei-Ge Chen ,  Naveen Thumpudi ,  Ming-Chieh Lee

发明人： Wei-Ge Chen ,  Naveen Thumpudi ,  Ming-Chieh Lee

IPC分类号： G10L19/00

CPC分类号： G10L25/69

摘要： An audio processing tool measures the quality of reconstructed audio data. For example, an audio encoder measures the quality of a block of reconstructed frequency coefficient data in a quantization loop. The invention includes several techniques and tools, which can be used in combination or separately. First, before measuring quality, the tool normalizes the block to account for variation in block sizes. Second, for the quality measurement, the tool processes the reconstructed data by critical bands, which can differ from the quantization bands used to compress the data. Third, the tool accounts for the masking effect of the reconstructed data, not just the masking effect of the original data. Fourth, the tool band weights the quality measurement, which can be used to account for noise substitution or band truncation. Finally, the tool changes quality measurement techniques depending on the channel coding mode.

公开(公告)号：US20060241796A1

公开(公告)日：2006-10-26

申请号：US11114873

申请日：2005-04-25

申请人： Christopher Messer ,  Naveen Thumpudi ,  Raymond Cheng ,  Serge Smirnov ,  Wei-ge Chen ,  Timothy Onders

发明人： Christopher Messer ,  Naveen Thumpudi ,  Raymond Cheng ,  Serge Smirnov ,  Wei-ge Chen ,  Timothy Onders

IPC分类号： G06F17/00

CPC分类号： G10L19/167 ,  H04N21/2368 ,  H04N21/4341 ,  H04N21/439

摘要： A compressed digital audio signal is transmitted from an audio source along a connection wire to an audio receiver. The digital audio signal can encode digital audio data having different sampling frequencies, frames sizes, and other information. The audio receiver that receives the digital audio signal can decode and convert the compressed digital audio signal into multiple synchronized analog signals, which are used to drive multiple speakers. The audio receiver may also synchronize the audio data with associated video data so that the audio playback and video playback are “in sync”, despite delay introduced by the audio signal decoding at the audio receiver.

摘要翻译： 压缩的数字音频信号从音频源沿着连接线传输到音频接收器。 数字音频信号可以编码具有不同采样频率，帧大小和其他信息的数字音频数据。 接收数字音频信号的音频接收器可以将压缩的数字音频信号解码并转换成多个同步的模拟信号，用于驱动多个扬声器。 音频接收器还可以使音频数据与相关联的视频数据同步，使音频播放和视频播放“尽管同步”，尽管由音频接收器处的音频信号解码引入了延迟。

公开(公告)号：US07092879B2

公开(公告)日：2006-08-15

申请号：US11169602

申请日：2005-06-28

申请人： Wei-Ge Chen ,  Ming-Chieh Lee

发明人： Wei-Ge Chen ,  Ming-Chieh Lee

IPC分类号： G10L19/00

CPC分类号： G10L19/173 ,  G10L19/032

摘要： A transcoder reduces excess requantization error in quantization of spectral data. The transcoder phase shifts data decompressed by a decompressor. The phase shifting causes a change to corresponding spectral data produced in later transform coding of the decompressed data. When the spectral data is then quantized to reduce bitrate, the earlier phase shifting reduces excess requantization error. After transcoding, a second decompressor can compensate for the phase shifting by, for example, reverse shifting by the amount of the phase shift. Instead of phase shifting, the transcoder can reduce excess requantization error by, for example, adding random noise to the decompressed data or changing transform block sizes.

公开(公告)号：US20050278172A1

公开(公告)日：2005-12-15

申请号：US10869467

申请日：2004-06-15

申请人： Kazuhito Koishida ,  Feng Zhuge ,  Hosam Khalil ,  Tian Wang ,  Wei-ge Chen

发明人： Kazuhito Koishida ,  Feng Zhuge ,  Hosam Khalil ,  Tian Wang ,  Wei-ge Chen

IPC分类号： G10L15/20 ,  G10L21/02

CPC分类号： G10L21/0208 ,  G10L21/0232

摘要： 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.

摘要翻译： 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。

公开(公告)号：US20050240398A1

公开(公告)日：2005-10-27

申请号：US11169602

申请日：2005-06-28

申请人： Wei-Ge Chen ,  Ming-Chieh Lee

发明人： Wei-Ge Chen ,  Ming-Chieh Lee

IPC分类号： G10L19/02 ,  G10L19/12 ,  G10L19/14

CPC分类号： G10L19/173 ,  G10L19/032

摘要： A transcoder reduces excess requantization error in quantization of spectral data. The transcoder phase shifts data decompressed by a decompressor. The phase shifting causes a change to corresponding spectral data produced in later transform coding of the decompressed data. When the spectral data is then quantized to reduce bitrate, the earlier phase shifting reduces excess requantization error. After transcoding, a second decompressor can compensate for the phase shifting by, for example, reverse shifting by the amount of the phase shift. Instead of phase shifting, the transcoder can reduce excess requantization error by, for example, adding random noise to the decompressed data or changing transform block sizes.

摘要翻译： 代码转换器减少频谱数据量化中的过量重新量化误差。 代码转换器相位移位由解压缩器解压缩的数据。 相移导致对解压缩数据的后续变换编码中产生的对应频谱数据的改变。 当频谱数据被量化以降低比特率时，较早的相移减少了过量的再量化误差。 在代码转换之后，第二解压缩器可以通过例如相移量的相移来补偿相移。 代码转换器不是通过例如向解压缩数据添加随机噪声或改变变换块大小来减少过量的重新排序错误。

公开(公告)号：US20050234731A1

公开(公告)日：2005-10-20

申请号：US10966443

申请日：2004-10-14

申请人： Sudheer Sirivara ,  James Johnston ,  Naveen Thumpudi ,  Wei-ge Chen ,  Sergey Smirnov ,  Chris Messer

发明人： Sudheer Sirivara ,  James Johnston ,  Naveen Thumpudi ,  Wei-ge Chen ,  Sergey Smirnov ,  Chris Messer

IPC分类号： G11B20/12 ,  G10L19/00 ,  G10L19/14 ,  G11B20/10 ,  H03M7/30 ,  G10L21/00

CPC分类号： G10L19/167

摘要： Described techniques and tools include techniques and tools for mapping digital media data (e.g., audio, video, still images, and/or text, among others) in a given format to a transport or file container format useful for encoding the data on optical disks such as digital video disks (DVDs). A digital media universal elementary stream can be used to map digital media streams (e.g., an audio stream, video stream or an image) into any arbitrary transport or file container, including optical disk formats, and other transports, such as broadcast streams, wireless transmissions, etc. The information to decode any given frame of the digital media in the stream can be carried in each coded frame. A digital media universal elementary stream includes stream components called chunks. An implementation of a digital media universal elementary stream arranges data for a media stream in frames, the frames having one or more chunks.

摘要翻译： 描述的技术和工具包括用于将给定格式的数字媒体数据（例如，音频，视频，静止图像和/或文本等）映射到用于对光盘上的数据进行编码有用的传输或文件容器格式的技术和工具 例如数字视频盘（DVD）。 数字媒体通用基本流可用于将数字媒体流（例如，音频流，视频流或图像）映射到任何任意的传输或文件容器中，包括光盘格式和其他传输，例如广播流，无线 传输等。用于解码流中数字媒体的任何给定帧的信息可以在每个编码帧中传送。 数字媒体通用基本流包括称为块的流组件。 数字媒体通用基本流的实现将帧中的媒体流的数据排列，帧具有一个或多个块。

公开(公告)号：US20050228651A1

公开(公告)日：2005-10-13

申请号：US10816466

申请日：2004-03-31

申请人： Tian Wang ,  Hosam Khalil ,  Kazuhito Koishida ,  Wei-Ge Chen ,  Mu Han

发明人： Tian Wang ,  Hosam Khalil ,  Kazuhito Koishida ,  Wei-Ge Chen ,  Mu Han

IPC分类号： G10L11/06 ,  G10L19/08

CPC分类号： G10L19/08 ,  G10L19/005 ,  G10L19/22

摘要： 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.

摘要翻译： 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。

公开(公告)号：US20050149324A1

公开(公告)日：2005-07-07

申请号：US11061012

申请日：2005-02-17

申请人： Wei-Ge Chen ,  Naveen Thumpudi ,  Ming-Chieh Lee

发明人： Wei-Ge Chen ,  Naveen Thumpudi ,  Ming-Chieh Lee

IPC分类号： G10L19/008 ,  G10L19/02 ,  G10L19/12

CPC分类号： G10L19/008 ,  G10L19/02 ,  G10L19/0204

摘要： Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode.

摘要翻译： 量化矩阵便于数字音频编码和解码。 音频编码器生成并压缩量化矩阵; 音频解码器解压缩并应用量化矩阵。 本发明包括可以组合或分开使用的几种技术和工具。 例如，音频编码器可以生成用于音频数据块的临界频带模式的量化矩阵。 编码器可以直接从临界频带模式计算量化矩阵，这可以从正被压缩的相同音频数据计算。 音频编码器/解码器可以根据多声道音频数据的编码信道模式使用不同的模式来产生/应用量化矩阵。 音频编码器/解码器可以对量化矩阵使用不同的压缩/解压缩模式，包括参数压缩/解压缩模式。

公开(公告)号：US20050143993A1

公开(公告)日：2005-06-30

申请号：US11067170

申请日：2005-02-24

申请人： Wei-Ge Chen ,  Naveen Thumpudi ,  Ming-Chieh Lee

发明人： Wei-Ge Chen ,  Naveen Thumpudi ,  Ming-Chieh Lee

IPC分类号： G10L19/002 ,  G10L19/24 ,  G10L19/00

CPC分类号： G10L19/24 ,  G10L19/002

摘要： An audio encoder regulates quality and bitrate with a control strategy. The strategy includes several features. First, an encoder regulates quantization using quality, minimum bit count, and maximum bit count parameters. Second, an encoder regulates quantization using a noise measure that indicates reliability of a complexity measure. Third, an encoder normalizes a control parameter value according to block size for a variable-size block. Fourth, an encoder uses a bit-count control loop de-linked from a quality control loop. Fifth, an encoder addresses non-monotonicity of quality measurement as a function of quantization level when selecting a quantization level. Sixth, an encoder uses particular interpolation rules to find a quantization level in a quality or bit-count control loop. Seventh, an encoder filters a control parameter value to smooth quality. Eighth, an encoder corrects model bias by adjusting a control parameter value in view of current buffer fullness.

公开(公告)号：US20050143991A1

公开(公告)日：2005-06-30

申请号：US11066898

申请日：2005-02-24

申请人： Wei-Ge Chen ,  Naveen Thumpudi ,  Ming-Chieh Lee

发明人： Wei-Ge Chen ,  Naveen Thumpudi ,  Ming-Chieh Lee

IPC分类号： G10L19/002 ,  G10L19/24 ,  G10L19/00

CPC分类号： G10L19/24 ,  G10L19/002

摘要： An audio encoder regulates quality and bitrate with a control strategy. The strategy includes several features. First, an encoder regulates quantization using quality, minimum bit count, and maximum bit count parameters. Second, an encoder regulates quantization using a noise measure that indicates reliability of a complexity measure. Third, an encoder normalizes a control parameter value according to block size for a variable-size block. Fourth, an encoder uses a bit-count control loop de-linked from a quality control loop. Fifth, an encoder addresses non-monotonicity of quality measurement as a function of quantization level when selecting a quantization level. Sixth, an encoder uses particular interpolation rules to find a quantization level in a quality or bit-count control loop. Seventh, an encoder filters a control parameter value to smooth quality. Eighth, an encoder corrects model bias by adjusting a control parameter value in view of current buffer fullness.