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1. 发明申请

US20090220110A1 SYSTEM AND METHOD OF REDUCING POWER CONSUMPTION FOR AUDIO PLAYBACK 有权
标题翻译：降低音频播放功耗的系统和方法
公开(公告)号：US20090220110A1
公开(公告)日：2009-09-03
申请号：US12041414
申请日：2008-03-03
申请人： Seyfollah Bazarjani , Guoqing Miao , Joseph R. Fitzgerald , Prajakt V. Kulkarni , Justin Joseph Rosen Gagne , Gene H. McAllister , Jeffrey Hinrichs , Jan Paul van der Wagt
发明人： Seyfollah Bazarjani , Guoqing Miao , Joseph R. Fitzgerald , Prajakt V. Kulkarni , Justin Joseph Rosen Gagne , Gene H. McAllister , Jeffrey Hinrichs , Jan Paul van der Wagt
IPC分类号： H03F21/00
CPC分类号： H03F1/0222 , H03F1/0211 , H03F2200/03 , H03G3/30 , H03G3/3005 , H03G3/32
摘要： A system and method of improving the efficiency in the power consumption of an audio system. In essence, the technique is to adjust the power delivered from the power supply to the analog section, such as the power amplifier, in response to the volume level indicated by the volume control module and/or in response to the detected characteristic of the input audio signal. Thus, in this manner, the analog section is operated in a manner that is related to the level of the signal it is processing. Additionally, the system and method also relate to a technique of adjusting the dynamic ranges of the digital signal and the analog signal to improve the overall dynamic range of the system without needing to consume additional power.
摘要翻译：一种提高音频系统功耗效率的系统和方法。实质上，该技术是根据由音量控制模块指示的音量水平和/或响应于检测到的输入特性来调节从电源传递到模拟部分（例如功率放大器）的功率音频信号。因此，以这种方式，模拟部分以与其正在处理的信号的电平相关的方式操作。此外，该系统和方法还涉及一种调整数字信号和模拟信号的动态范围的技术，以改善系统的整体动态范围，而不需要消耗附加功率。

2. 发明授权

US09425747B2 System and method of reducing power consumption for audio playback 有权
标题翻译：降低音频播放功耗的系统和方法
公开(公告)号：US09425747B2
公开(公告)日：2016-08-23
申请号：US12041414
申请日：2008-03-03
申请人： Seyfollah Bazarjani , Guoqing Miao , Joseph R. Fitzgerald , Prajakt V. Kulkarni , Justin Joseph Rosen Gagne , Gene H. McAllister , Jeffrey Hinrichs , Jan Paul van der Wagt
发明人： Seyfollah Bazarjani , Guoqing Miao , Joseph R. Fitzgerald , Prajakt V. Kulkarni , Justin Joseph Rosen Gagne , Gene H. McAllister , Jeffrey Hinrichs , Jan Paul van der Wagt
IPC分类号： H03F99/00 , H03G3/00 , H03F1/02 , H03G3/30 , H03G3/32
CPC分类号： H03F1/0222 , H03F1/0211 , H03F2200/03 , H03G3/30 , H03G3/3005 , H03G3/32
摘要： A system and method of improving the efficiency in the power consumption of an audio system. In essence, the technique is to adjust the power delivered from the power supply to the analog section, such as the power amplifier, in response to the volume level indicated by the volume control module and/or in response to the detected characteristic of the input audio signal. Thus, in this manner, the analog section is operated in a manner that is related to the level of the signal it is processing. Additionally, the system and method also relate to a technique of adjusting the dynamic ranges of the digital signal and the analog signal to improve the overall dynamic range of the system without needing to consume additional power.
摘要翻译：一种提高音频系统功耗效率的系统和方法。实质上，该技术是根据由音量控制模块指示的音量水平和/或响应于检测到的输入特性来调节从电源传递到模拟部分（例如功率放大器）的功率音频信号。因此，以这种方式，模拟部分以与其正在处理的信号的电平相关的方式操作。此外，该系统和方法还涉及一种调整数字信号和模拟信号的动态范围的技术，以改善系统的整体动态范围，而不需要消耗附加功率。

3. 发明申请

US20080229911A1 WAVEFORM FETCH UNIT FOR PROCESSING AUDIO FILES 有权
标题翻译：用于处理音频文件的波形FETCH单元
公开(公告)号：US20080229911A1
公开(公告)日：2008-09-25
申请号：US12041834
申请日：2008-03-04
申请人： Nidish Ramachandra Kamath , Prajakt V. Kulkarni , Samir Kumar Gupta , Stephen Molloy , Suresh Devalapalli , Allister Alemania
发明人： Nidish Ramachandra Kamath , Prajakt V. Kulkarni , Samir Kumar Gupta , Stephen Molloy , Suresh Devalapalli , Allister Alemania
IPC分类号： G10H7/00
CPC分类号： G10H7/004 , G10H2230/031 , G10H2250/641
摘要： This disclosure describes techniques that make use of a waveform fetch unit that operates to retrieve waveform samples on behalf of each of a plurality of hardware processing elements that operate simultaneously to service various audio synthesis parameters generated from one or more audio files, such as musical instrument digital interface (MIDI) files. In one example, a method comprises receiving a request for a waveform sample from an audio processing element, and servicing the request by calculating a waveform sample number for the requested waveform sample based on a phase increment contained in the request and an audio synthesis parameter control word associated with the requested waveform sample, retrieving the waveform sample from a local cache using the waveform sample number, and sending the retrieved waveform sample to the requesting audio processing element.
摘要翻译：本公开描述了利用波形获取单元的技术，其操作以代表同时操作的多个硬件处理元件中的每一个来检索波形采样，以服务从一个或多个音频文件（例如乐器）生成的各种音频合成参数数字接口（MIDI）文件。在一个示例中，一种方法包括从音频处理元件接收对波形样本的请求，并且通过基于包含在请求中的相位增量和音频合成参数控制来计算所请求的波形样本的波形采样数来为该请求提供服务与所请求的波形样本相关联的单词，使用波形采样号从本地高速缓存中检索波形采样，以及将检索的波形样本发送到请求音频处理单元。

4. 发明申请

US20070286426A1 Mixing techniques for mixing audio 有权
标题翻译：混音技术混音
公开(公告)号：US20070286426A1
公开(公告)日：2007-12-13
申请号：US11449454
申请日：2006-06-07
申请人： Pei Xiang , Eddie L.T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
发明人： Pei Xiang , Eddie L.T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
IPC分类号： H04R5/00
CPC分类号： H04S7/30 , H04M1/6016 , H04M1/72558 , H04R2499/11
摘要： This disclosure describes audio mixing techniques that intelligently combine two or more audio signals into an output signal. The techniques allow audio to be combined, yet create perceptual differentiation between the different audio signals. The result is that a user is able to hear both audio signals in a combined output, but the different audio signals do not perceptually interfere with one another. The techniques are relatively simple to implement and are well suited for radio telephones.
摘要翻译：本公开描述了将两个或多个音频信号智能地组合成输出信号的音频混合技术。这些技术允许音频被组合，但是在不同音频信号之间产生感知差异。结果是用户能够听到组合输出中的两个音频信号，但是不同的音频信号不会感知到彼此干扰。这些技术实现起来相对简单，并且非常适合于无线电话。

5. 发明授权

US07742746B2 Automatic volume and dynamic range adjustment for mobile audio devices 有权
标题翻译：移动音频设备的自动音量和动态范围调整
公开(公告)号：US07742746B2
公开(公告)日：2010-06-22
申请号：US11742476
申请日：2007-04-30
申请人： Pei Xiang , Song Wang , Prajakt V. Kulkarni , Samir Kumar Gupta , Eddie L. T. Choy
发明人： Pei Xiang , Song Wang , Prajakt V. Kulkarni , Samir Kumar Gupta , Eddie L. T. Choy
IPC分类号： H04B1/00
CPC分类号： H03G7/007 , H03G3/32 , H04M1/6016
摘要： A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.
摘要翻译：移动音频设备（例如，蜂窝电话，个人数字音频播放器或MP3播放器）执行音频动态范围控制（ADRC）和自动音量控制（AVC）以增加从移动音频的扬声器发出的声音的音量设备使得音频的微弱通道更可听见。这种微弱通道的放大发生，而不会过度放大其他更大的通道，并且没有由于限幅导致的实质性变形。例如，多麦克风有源噪声消除（MMANC）功能用于从移动音频设备的麦克风拾取的音频信息中去除背景噪声。然后可以从设备传送噪声消除的音频。 MMANC功能产生噪声参考信号作为中间信号。中间信号被调节，然后用作AVC处理的参考。在AVC过程中应用的增益是噪声参考信号的函数。

6. 发明申请

US20090086982A1 CROSSTALK CANCELLATION FOR CLOSELY SPACED SPEAKERS 审中-公开
标题翻译：闭路遥控扬声器的CROSSTALK取消
公开(公告)号：US20090086982A1
公开(公告)日：2009-04-02
申请号：US11864552
申请日：2007-09-28
申请人： Prajakt V. Kulkarni , Pei Xiang
发明人： Prajakt V. Kulkarni , Pei Xiang
IPC分类号： H04R5/00
CPC分类号： H04R5/04 , H04S1/002
摘要： A technique for canceling acoustic crosstalk is provided including a pre-processing filter and a crosstalk cancellation device. The pre-processing filter may be configured to obtain first and second channel signals and compensate or adjust the first and/or second channel signals for anticipated subsequent stage distortion by the crosstalk cancellation device. The crosstalk cancellation device maybe configured to receive the compensated first and second channel signals from the pre-processing filter. The crosstalk cancellation device then modifies the first channel signal to cancel anticipated acoustic crosstalk from the second channel signal, and modifies the second channel signal to cancel acoustic crosstalk from the first channel signal. The modified first channel signal is then transmitted over a first speaker and the modified second channel signal is transmitted over a second speaker. The first and second speakers may be closely spaced, yet provide a widened stereo image of the first and second channel signals.
摘要翻译：提供一种用于消除声学串扰的技术，包括预处理滤波器和串扰消除装置。预处理滤波器可以被配置为获得第一和第二信道信号，并且通过串扰消除装置来补偿或调整第一和/或第二信道信号以用于预期的后级失真。串扰消除装置可以被配置为从预处理滤波器接收经补偿的第一和第二信道信号。然后，串扰消除装置修改第一信道信号以从第二信道信号中消除预期的声学串扰，并且修改第二信道信号以从第一信道信号中消除声学串扰。经修改的第一信道信号然后通过第一扬声器发送，并且修改的第二信道信号通过第二扬声器发送。第一和第二扬声器可以是紧密间隔的，但是提供第一和第二声道信号的加宽立体图像。

7. 发明申请

US20080269926A1 AUTOMATIC VOLUME AND DYNAMIC RANGE ADJUSTMENT FOR MOBILE AUDIO DEVICES 有权
标题翻译：自动音量和动态范围调整移动音频设备
公开(公告)号：US20080269926A1
公开(公告)日：2008-10-30
申请号：US11742476
申请日：2007-04-30
申请人： Pei Xiang , Song Wang , Prajakt V. Kulkarni , Samir Kumar Gupta , Eddie L.T. Choy
发明人： Pei Xiang , Song Wang , Prajakt V. Kulkarni , Samir Kumar Gupta , Eddie L.T. Choy
IPC分类号： H04S7/00
CPC分类号： H03G7/007 , H03G3/32 , H04M1/6016
摘要： A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.
摘要翻译：移动音频设备（例如，蜂窝电话，个人数字音频播放器或MP3播放器）执行音频动态范围控制（ADRC）和自动音量控制（AVC）以增加从移动音频的扬声器发出的声音的音量设备使得音频的微弱通道更可听见。这种微弱通道的放大发生，而不会过度放大其他更大的通道，并且没有由于限幅导致的实质性变形。例如，多麦克风有源噪声消除（MMANC）功能用于从移动音频设备的麦克风拾取的音频信息中去除背景噪声。然后可以从设备传送噪声消除的音频。 MMANC功能产生噪声参考信号作为中间信号。中间信号被调节，然后用作AVC处理的参考。在AVC过程中应用的增益是噪声参考信号的函数。

8. 发明申请

US20090136044A1 METHODS AND APPARATUS FOR PROVIDING A DISTINCT PERCEPTUAL LOCATION FOR AN AUDIO SOURCE WITHIN AN AUDIO MIXTURE 有权
标题翻译：用于在音频混合中提供音频源的异常识别位置的方法和装置
公开(公告)号：US20090136044A1
公开(公告)日：2009-05-28
申请号：US11946365
申请日：2007-11-28
申请人： Pei Xiang , Samir Kumar Gupta , Eddie L. T. Choy , Prajakt V. Kulkarni
发明人： Pei Xiang , Samir Kumar Gupta , Eddie L. T. Choy , Prajakt V. Kulkarni
IPC分类号： H04H20/00
CPC分类号： H04S7/30 , H04S7/302 , H04S2400/11
摘要： In accordance with a method for providing a distinct perceptual location for an audio source within an audio mixture, a foreground signal may be processed to provide a foreground perceptual angle for the foreground signal. The foreground signal may also be processed to provide a desired attenuation level for the foreground signal. A background signal may be processed to provide a background perceptual angle for the background signal. The background signal may also be processed to provide a desired attenuation level for the background signal. The foreground signal and the background signal may be combined into an output audio source.
摘要翻译：根据用于为音频混合中的音频源提供不同的感知位置的方法，可以处理前景信号以提供前景信号的前景感知角度。还可以处理前景信号以为前景信号提供期望的衰减电平。可以处理背景信号以为背景信号提供背景感知角度。背景信号也可以被处理以为背景信号提供期望的衰减水平。前景信号和背景信号可以组合成输出音频源。

9. 发明授权

US07528745B2 Digital domain sampling rate converter 有权
标题翻译：数字域采样率转换器
公开(公告)号：US07528745B2
公开(公告)日：2009-05-05
申请号：US11452836
申请日：2006-06-13
申请人： Song Wang , Eddie L. T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
发明人： Song Wang , Eddie L. T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
IPC分类号： H03M7/00
CPC分类号： H03H17/0685 , H03H17/0294
摘要： Techniques are described for sampling rate conversion in the digital domain by up-sampling and down-sampling a digital signal according to a selected intermediate sampling frequency. A prototype anti-aliasing filter that has a bandwidth with multiple factors is stored in memory. The techniques include selecting an intermediate sampling frequency to be an integer multiple of a desired output sampling frequency of a digital signal based on the factors of the prototype filter, and selecting a down-sampling factor to be the same integer associated with the selected intermediate sampling frequency. A filter generator generates an anti-aliasing filter for the selected down-sampling factor based on the prototype filter. A sampling rate converter up-samples the digital signal at an input sampling frequency to the selected intermediate sampling frequency, filters the digital signal with the derived anti-aliasing filter, and down-samples the digital signal by the selected down-sampling factor to the desired output sampling frequency.
摘要翻译：描述了通过根据所选择的中间采样频率对数字信号进行上采样和下采样来对数字域中的采样率转换进行描述的技术。具有多个因素的带宽的原型抗混叠滤波器存储在存储器中。这些技术包括基于原型滤波器的因素来选择中间采样频率为数字信号的期望输出采样频率的整数倍，并且将下采样因子选择为与所选择的中间采样相关联的整数频率。滤波器发生器基于原型滤波器生成用于所选择的下采样因子的抗混叠滤波器。采样率转换器将数字信号以输入采样频率向采样频率进行上采样，以采样导出的抗混叠滤波器对数字信号进行滤波，并通过选择的下采样因子将数字信号下采样到所需输出采样频率。

10. 发明申请

US20070192390A1 Digital domain sampling rate converter 有权
标题翻译：数字域采样率转换器
公开(公告)号：US20070192390A1
公开(公告)日：2007-08-16
申请号：US11452836
申请日：2006-06-13
申请人： Song Wang , Eddie L.T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
发明人： Song Wang , Eddie L.T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
IPC分类号： G06F1/02
CPC分类号： H03H17/0685 , H03H17/0294
摘要： Techniques are described for sampling rate conversion in the digital domain by up-sampling and down-sampling a digital signal according to a selected intermediate sampling frequency. A prototype anti-aliasing filter that has a bandwidth with multiple factors is stored in memory. The techniques include selecting an intermediate sampling frequency to be an integer multiple of a desired output sampling frequency of a digital signal based on the factors of the prototype filter, and selecting a down-sampling factor to be the same integer associated with the selected intermediate sampling frequency. A filter generator generates an anti-aliasing filter for the selected down-sampling factor based on the prototype filter. A sampling rate converter up-samples the digital signal at an input sampling frequency to the selected intermediate sampling frequency, filters the digital signal with the derived anti-aliasing filter, and down-samples the digital signal by the selected down-sampling factor to the desired output sampling frequency.
摘要翻译：描述了通过根据所选择的中间采样频率对数字信号进行上采样和下采样来对数字域中的采样率转换进行描述的技术。具有多个因素的带宽的原型抗混叠滤波器存储在存储器中。这些技术包括基于原型滤波器的因素来选择中间采样频率为数字信号的期望输出采样频率的整数倍，并且将下采样因子选择为与所选择的中间采样相关联的整数频率。滤波器发生器基于原型滤波器生成用于所选择的下采样因子的抗混叠滤波器。采样率转换器将数字信号以输入采样频率向采样频率进行上采样，以采样导出的抗混叠滤波器对数字信号进行滤波，并通过选择的下采样因子将数字信号下采样到所需输出采样频率。

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