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公开(公告)号：ES2269112T3
公开(公告)日：2007-04-01
申请号：ES00912053
申请日：2000-02-29
Applicant: QUALCOMM INC
Inventor： DAS AMITAVA
Abstract: Procedimiento para procesar tramas, que comprende las etapas siguientes: aplicar un procedimiento de selección de modalidad de codificación en bucle abierto a cada trama de entrada consecutiva para seleccionar una modalidad de codificación en el dominio del tiempo o una modalidad de codificación en el dominio de la frecuencia, basándose en el contenido de voz de la trama de entrada; codificar en el dominio de la frecuencia (416) la trama de entrada si el contenido de voz de la trama de entrada indica voz sonora de estado estacionario; codificar en el dominio del tiempo (414) la trama de entrada si el contenido de voz de la trama de entrada indica cualquier tipo de voz que no es voz sonora de estado estacionario; comparar (420) la trama codificada en el dominio de la frecuencia con la trama de entrada para obtener una medida del rendimiento; y codificar en el dominio del tiempo (414) la trama de entrada si la medida del rendimiento está por debajo de un valor umbral predefinido.
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公开(公告)号：ES2253226T3
公开(公告)日：2006-06-01
申请号：ES00930512
申请日：2000-05-08
Applicant: QUALCOMM INC
Inventor： DAS AMITAVA , MANJUNATH SHARATH
Abstract: Un método para codificar tramas transicionales de voz que incluye los pasos de: representar una primera trama de muestras de voz transicional mediante un primer subconjunto de muestras (402, 416) de la primera trama ; e interpolar (420) el primer subconjunto de muestras (402, 416) y un segundo subconjunto alineado de muestras (418) extraído de una segunda trama anteriormente recibida de muestras de voz transicional para sintetizar otras muestras de la primera trama que no están incluidas en el primer subconjunto.
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公开(公告)号：DE69924280D1
公开(公告)日：2005-04-21
申请号：DE69924280
申请日：1999-11-12
Applicant: QUALCOMM INC
Inventor： DAS AMITAVA , CHOY L
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.
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公开(公告)号：AT286617T
公开(公告)日：2005-01-15
申请号：AT99958940
申请日：1999-11-12
Applicant: QUALCOMM INC
Inventor： DAS AMITAVA , MANJUNATH SHARATH
Abstract: A low-bit-rate coding technique for unvoiced segments of speech includes the steps of extracting high-time-resolution energy coefficients from a frame of speech, quantizing the energy coefficients, generating a high-time-resolution energy envelope from the quantized energy coefficients, and reconstituting a residue signal by shaping a randomly generated noise vector with quantized values of the energy envelope. The energy envelope may be generated with a linear interpolation technique. A post-processing measure may be obtained and compared with a predefined threshold to determine whether the coding algorithm is performing adequately.
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公开(公告)号：DE60031002T2
公开(公告)日：2007-05-10
申请号：DE60031002
申请日：2000-02-29
Applicant: QUALCOMM INC
Inventor： DAS AMITAVA
Abstract: A closed-loop, multimode, mixed-domain linear prediction (MDLP) speech coder includes a high-rate, time-domain coding mode, a low-rate, frequency-domain coding mode, and a closed-loop mode-selection mechanism for selecting a coding mode for the coder based upon the speech content of frames input to the coder. Transition speech (i.e., from unvoiced speech to voiced speech, or vice versa) frames are encoded with the high-rate, time-domain coding mode, which may be a CELP coding mode. Voiced speech frames are encoded with the low-rate, frequency-domain coding mode, which may be a harmonic coding mode. Phase parameters are not encoded by the frequency-domain coding mode, and are instead modeled in accordance with, e.g., a quadratic phase model. For each speech frame encoded with the frequency-domain coding mode, the initial phase value is taken to be the initial phase value of the immediately preceding speech frame encoded with the frequency-domain coding mode. If the immediately preceding speech frame was encoded with the time-domain coding mode, the initial phase value of the current speech frame is computed from the decoded speech frame information of the immediately preceding, time-domain-encoded speech frame. Each speech frame encoded with the frequency-domain coding mode may be compared with the corresponding input speech frame to obtain a performance measure. If the performance measure falls below a predefined threshold value, the input speech frame is encoded with the time-domain coding mode.
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Patent Agency Ranking
公开(公告)号：DE60031002D1
公开(公告)日：2006-11-09
申请号：DE60031002
申请日：2000-02-29
Applicant: QUALCOMM INC
Inventor： DAS AMITAVA
Abstract: A closed-loop, multimode, mixed-domain linear prediction (MDLP) speech coder includes a high-rate, time-domain coding mode, a low-rate, frequency-domain coding mode, and a closed-loop mode-selection mechanism for selecting a coding mode for the coder based upon the speech content of frames input to the coder. Transition speech (i.e., from unvoiced speech to voiced speech, or vice versa) frames are encoded with the high-rate, time-domain coding mode, which may be a CELP coding mode. Voiced speech frames are encoded with the low-rate, frequency-domain coding mode, which may be a harmonic coding mode. Phase parameters are not encoded by the frequency-domain coding mode, and are instead modeled in accordance with, e.g., a quadratic phase model. For each speech frame encoded with the frequency-domain coding mode, the initial phase value is taken to be the initial phase value of the immediately preceding speech frame encoded with the frequency-domain coding mode. If the immediately preceding speech frame was encoded with the time-domain coding mode, the initial phase value of the current speech frame is computed from the decoded speech frame information of the immediately preceding, time-domain-encoded speech frame. Each speech frame encoded with the frequency-domain coding mode may be compared with the corresponding input speech frame to obtain a performance measure. If the performance measure falls below a predefined threshold value, the input speech frame is encoded with the time-domain coding mode.
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公开(公告)号：DE60025471T2
公开(公告)日：2006-08-24
申请号：DE60025471
申请日：2000-02-29
Applicant: QUALCOMM INC
Inventor： DAS AMITAVA
IPC: G10L19/005 , G10L19/02 , G10L19/04 , G10L25/48
Abstract: A method for tracking the phase of a quasi-periodic signal includes he steps of estimating the phase of the signal for frames during which the signal is periodic, monitoring the performance of the estimated phase with a closed-loop performance measure, and measuring the phase of the signal for frames during which the signal is periodic and performance of the estimated phase falls below a predefined threshold level. In estimating the phase, the initial phase value is set equal to the estimated final phase value of the previous frame if the previous frame was periodic. The initial phase value is set equal to a measured phase value of the previous frame if the previous frame was nonperiodic, or if the previous frame was periodic and performance of the estimated phase for the previous frame fell below the predefined threshold level. For frames during which the signal is nonperiodic, the phase of the signal is measured. An open-loop periodicity decision can be used to determine whether the signal is periodic for a given frame.
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公开(公告)号：HK1044614A1
公开(公告)日：2002-10-25
申请号：HK02106115
申请日：2002-08-21
Applicant: QUALCOMM INC
Inventor： DAS AMITAVA , MANJUNATH SHARATH
Abstract: A multipulse interpolative coder for transition speech frames includes an extractor configured to represent a first frame of transitional speech samples by a subset of the samples of the frame. The coder also includes an interpolator configured to interpolate the subset of samples and a subset of samples extracted from an earlier-received frame to synthesize other samples of the first frame that are not included in the subset. The subset of samples is further simplified by selecting a set of pulses from the subset and assigning zero values to unselected pulses. In the alternative, a portion of the unselected pulses may be quantized. The set of pulses may be the pulses having the greatest absolute amplitudes in the subset. In the alternative, the set of pulses may be the most perceptually significant pulses of the subset.
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公开(公告)号：AU3385100A
公开(公告)日：2001-09-12
申请号：AU3385100
申请日：2000-02-29
Applicant: QUALCOMM INC
Inventor： DAS AMITAVA
Abstract: A closed-loop, multimode, mixed-domain linear prediction (MDLP) speech coder includes a high-rate, time-domain coding mode, a low-rate, frequency-domain coding mode, and a closed-loop mode-selection mechanism for selecting a coding mode for the coder based upon the speech content of frames input to the coder. Transition speech (i.e., from unvoiced speech to voiced speech, or vice versa) frames are encoded with the high-rate, time-domain coding mode, which may be a CELP coding mode. Voiced speech frames are encoded with the low-rate, frequency-domain coding mode, which may be a harmonic coding mode. Phase parameters are not encoded by the frequency-domain coding mode, and are instead modeled in accordance with, e.g., a quadratic phase model. For each speech frame encoded with the frequency-domain coding mode, the initial phase value is taken to be the initial phase value of the immediately preceding speech frame encoded with the frequency-domain coding mode. If the immediately preceding speech frame was encoded with the time-domain coding mode, the initial phase value of the current speech frame is computed from the decoded speech frame information of the immediately preceding, time-domain-encoded speech frame. Each speech frame encoded with the frequency-domain coding mode may be compared with the corresponding input speech frame to obtain a performance measure. If the performance measure falls below a predefined threshold value, the input speech frame is encoded with the time-domain coding mode.
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10.发明专利
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
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.
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