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    MULTIPLE MICROPHONE VOICE ACTIVITY DETECTOR
    1.
    发明专利
    MULTIPLE MICROPHONE VOICE ACTIVITY DETECTOR 未知

    公开(公告)号：CA2695231C

    公开(公告)日：2015-02-17

    申请号：CA2695231

    申请日：2008-09-26

    Applicant: QUALCOMM INC

    Inventor： WANG SONG ,  GUPTA SAMIR KUMAR ,  CHOY EDDIE L T

    IPC: G10L25/78 ,  G10L21/0208

    Abstract: Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a prede-termined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

    MULTIPLE MICROPHONE VOICE ACTIVITY DETECTOR
    3.
    发明专利
    MULTIPLE MICROPHONE VOICE ACTIVITY DETECTOR 未知

    公开(公告)号：CA2695231A1

    公开(公告)日：2009-04-02

    申请号：CA2695231

    申请日：2008-09-26

    Applicant: QUALCOMM INC

    Inventor： WANG SONG ,  GUPTA SAMIR KUMAR ,  CHOY EDDIE L T

    IPC: G10L11/02

    Abstract: Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

    Synthesis of speech from pitch prototype waveforms by time-synchronous waveform interpolation
    5.
    发明专利
    Synthesis of speech from pitch prototype waveforms by time-synchronous waveform interpolation 未知

    公开(公告)号：AU1721100A

    公开(公告)日：2000-06-05

    申请号：AU1721100

    申请日：1999-11-12

    Applicant: QUALCOMM INC

    Inventor： DAS AMITAVA ,  CHOY EDDIE L T

    IPC: G10L19/04 ,  G10L13/00 ,  G10L19/00 ,  G10L19/02 ,  G10L19/12 ,  G10L25/90

    Abstract: In a method of synthesizing voiced speech from pitch prototype waveforms by time-synchronous waveform interpolation (TSWI), one or more pitch prototypes is extracted from a speech signal or a residue signal. The extraction process is performed in such a way that the prototype has minimum energy at the boundary. Each prototype is circularly shifted so as to be time-synchronous with the original signal. A linear phase shift is applied to each extracted prototype relative to the previously extracted prototype so as to maximize the cross-correlation between successive extracted prototypes. A two-dimensional prototype-evolving surface is constructed by unsampling the prototypes to every sample point. The two-dimensional prototype-evolving surface is re-sampled to generate a one-dimensional, synthesized signal frame with sample points defined by piecewise continuous cubic phase contour functions computed from the pitch lags and the phase shifts added to the extracted prototypes. A pre-selection filter may be applied to determine whether to abandon the TSWI technique in favor of another algorithm for the current frame. A post-selection performance measure may be obtained and compared with a predetermined threshold to determine whether the TSWI algorithm is performing adequately.

    SIGNALING MICROPHONE COVERING TO THE USER
    7.
    发明专利
    SIGNALING MICROPHONE COVERING TO THE USER 未知

    公开(公告)号：CA2705805A1

    公开(公告)日：2009-08-06

    申请号：CA2705805

    申请日：2009-01-29

    Applicant: QUALCOMM INC

    Inventor： RAMAKRISHNAN DINESH ,  WANG SONG ,  CHOY EDDIE L T ,  SATYANARAYANAN RAVI

    IPC: H04R29/00 ,  H04R3/00

    Abstract: A mechanism is provided that monitors secondary microphone signals, in a multi-microphone mobile device, to warn the user if one or more secondary microphones are covered while the mobile device is in use. In one example, smoothly averaged power estimates of the secondary microphones may be computed and compared against the noise floor estimate of a primary microphone. Microphone covering detection may be made by comparing the secondary microphone smooth power estimates to the noise floor estimate for the primary microphone. In another example, the noise floor estimates for the primary and secondary microphone signals may be compared to the difference in the sensitivity of the first and second microphones to determine if the secondary microphone is covered. Once detection is made, a warning signal may be generated and issued to the user.

    METHODS AND APPARATUS FOR PROVIDING A DISTINCT PERCEPTUAL LOCATION FOR AN AUDIO SOURCE WITHIN AN AUDIO MIXTURE
    8.
    发明专利
    METHODS AND APPARATUS FOR PROVIDING A DISTINCT PERCEPTUAL LOCATION FOR AN AUDIO SOURCE WITHIN AN AUDIO MIXTURE 未知

    公开(公告)号：CA2705776A1

    公开(公告)日：2009-06-04

    申请号：CA2705776

    申请日：2008-11-26

    Applicant: QUALCOMM INC

    Inventor： XIANG PEI ,  GUPTA SAMIR KUMAR ,  CHOY EDDIE L T ,  KULKARNI PRAJAKT V

    IPC: H04S7/00

    Abstract: 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.

    POWER EFFICIENT BATCH-FRAME AUDIO DECODING APPARATUS, SYSTEM AND METHOD
    9.
    发明申请
    POWER EFFICIENT BATCH-FRAME AUDIO DECODING APPARATUS, SYSTEM AND METHOD 审中-公开
    功率有效的批量音频解码设备，系统和方法

    公开(公告)号：WO2009033147A3

    公开(公告)日：2009-05-28

    申请号：PCT/US2008075578

    申请日：2008-09-08

    Applicant: QUALCOMM INC ,  YOO HEEJONG ,  KAMATH NIDISH RAMACHANDRA ,  CHOY EDDIE L T ,  JOHN JOHNNY KALLACHERIL ,  GUPTA SAMIR KUMAR

    Inventor： YOO HEEJONG ,  KAMATH NIDISH RAMACHANDRA ,  CHOY EDDIE L T ,  JOHN JOHNNY KALLACHERIL ,  GUPTA SAMIR KUMAR

    IPC: G06F1/32 ,  G06F3/16 ,  G06F5/06 ,  G10L19/14 ,  H04W52/02

    CPC classification number: G10L19/16 ,  G06F1/3203 ,  G06F3/162 ,  H04W52/0274 ,  Y02D70/00

    Abstract: Power savings in a mobile device is accomplished by generating audio samples by decoding a bitstream with a decoding system within the mobile device. The generated audio samples are transferred into at least one memory bank in a set of memory banks in a power saver block within the mobile device. Parts of the decoding system not involved in the storing of the generated audio samples are switched off after batch decoding a bitstream associated with multiple audio frames. The bitstream includes bits less than that found in one audio file. At least one of the memory banks in the set of memory banks is power collapsible. The fetching of the decoded by the decoding system can be synchronized with a paging channel of a modem in the mobile device. The transferred audio samples is a lossless compression and may occur after a re-encoding.

    Abstract translation: 移动设备中的功率节省通过利用移动设备内的解码系统对比特流进行解码来生成音频样本来实现。 生成的音频样本被传送到移动设备内的节电块中的一组存储器组中的至少一个存储体。 在对与多个音频帧相关联的比特流进行批量解码之后，不涉及生成的音频样本的存储的部分解码系统被关闭。 比特流包括比在一个音频文件中发现的比特小的比特。 存储器组中的至少一个存储体是电源可折叠的。 由解码系统解码的提取可以与移动设备中的调制解调器的寻呼信道同步。 传输的音频样本是无损压缩，并且可能在重新编码之后发生。

    INTEGER REPRESENATION OF RELATIVE TIMING BETWEEN DESIRED OUTPUT SAMPLES AND CORRESPONDING INPUT SAMPLES
    10.
    发明申请
    INTEGER REPRESENATION OF RELATIVE TIMING BETWEEN DESIRED OUTPUT SAMPLES AND CORRESPONDING INPUT SAMPLES 审中-公开
    所有输出样本和相关输入样本之间相对时间的整体表示

    公开(公告)号：WO2007146599A2

    公开(公告)日：2007-12-21

    申请号：PCT/US2007070038

    申请日：2007-05-31

    Applicant: QUALCOMM INC ,  WANG SONG ,  CHOY EDDIE L T ,  GUPTA SAMIR KUMAR

    Inventor： WANG SONG ,  CHOY EDDIE L T ,  GUPTA SAMIR KUMAR

    IPC: H03H17/06

    CPC classification number: H03H17/0685

    Abstract: In general, this disclosure describes techniques for changing a sampling frequency of a digital signal. In particular, the techniques provide a more accurate way to determining a relative timing between a desired output sample and a corresponding input sample using a non-approximated integer representation of the relative timing. The relative timing between the desired output sample and corresponding input sample may be represented using a first component that identifies a latest input sample of the digital signal used to generate intermediate samples, a second component that identifies an intermediate sample, and a third component that identifies a timing difference between the desired output sample and the intermediate sample. Each of the components may be recursively updated using non-approximated integer values.

    Abstract translation: 通常，本公开描述了用于改变数字信号的采样频率的技术。 特别地，这些技术提供了使用相对定时的非近似整数表示来确定期望输出采样和相应输入采样之间的相对定时的更精确的方法。 可以使用标识用于生成中间样本的数字信号的最新输入样本的第一组件，标识中间样本的第二组件和标识中间样本的第三组件来表示期望输出样本与相应输入样本之间的相对时序 所需输出样本和中间样本之间的时间差。 可以使用非近似的整数值递归地更新每个组件。

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