摘要:
There is provided a speech encoding system that receives a speech signal. The speech encoding system comprises a frame processor for processing a frame of the speech signal, where the frame processor includes a pitch gain generator that derives unquantized pitch gains, and a first vector guantizer that receives the unquantized pitch gains and generates quantized pitch gains. The speech encoding system also comprises a subframe processor that begins subframe processing after the pitch gain generator has derived the unquantized pitch gains and the first vector quantizer has generated the quantized pitch gains.
摘要:
The invention provides Aryl substituted imidazoles, pyrazoles, pyridizines and related compounds of the Formula where the variables are defined herein. Such compounds are ligands of C5a receptors. Preferred compounds of the invention act bind to C5a receptors with high affinity and exhibit neutral antagonist or inverse agonist activity at C5a receptos. This invention also relates to pharmaceutical compositions comprising such compounds. It further relates to the use of such compounds in treating a variety of inflammatory and immune system disorders.
摘要:
There is provided a voice activity detection method for indicating an active voice mode and an inactive voice mode. The method comprises receiving a first portion of an input signal; determining that the first portion of the input signal includes an active voice signal; indicating the active voice mode in response to the determining that the first portion of the input signal includes the active voice signal; receiving a second portion of the input signal immediately following the first portion of the input signal; determining that the second portion of the input signal includes an inactive voice signal; extending the indicating the active voice mode for a period of time after determining that the second portion of the input signal includes the inactive voice signal, wherein the period of time varies based on one or more conditions; and indicating the inactive voice mode after expiration of the period of time.
摘要:
A system including an adaptive code book and a fixed code book for code excited linear prediction coding of speech signals is provided. The invention includes an embedded adaptive code book in the fixed code book and the selection procedure for selecting excitation vector parameters. A code book update system updates the fixed code book with embedded adaptive code book based on the long term processing excitation vector parameters with previous synthesized excitation.
摘要:
Disclosed are compounds for SH2 domain binding inhibition. For example, disclosed is a compound of formula (I) wherein R1 is a lipophile; R2, in combination with the phenyl ring, forms a phenylphosphate mimic group or a protected phenylphosphate mimic group; R3 is hydrogen, azido, amino, carboxyalkyl, alkoxycarbonylalkyl, aminocarbonylalkyl, or alkylcarbonylamino, wherein the alkyl portion of R3 may be optionally substituted with a substituent selected from the group consisting of halo, hydroxy, carboxyl, amino, aminoalkyl, alkyl, alkoxy, and keto; R6 is a linker; AA is an amino acid; and n is 1 to 6; or a salt thereof. Also disclosed are a pharmaceutical composition, a method for inhibiting an SH2 domain from binding with a phosphoprotein and a method of treating breast cancer.
摘要翻译:公开了用于SH2结构域结合抑制的化合物。 例如,公开了式(I)的化合物,其中R 1是亲脂体; R 2与苯环组合形成苯基磷酸酯模拟基团或被保护的苯基磷酸酯模拟基团; R 3是氢,叠氮基,氨基,羧基烷基,烷氧基羰基烷基,氨基羰基烷基或烷基羰基氨基,其中R 3的烷基部分可以任选地被选自 由卤素,羟基,羧基,氨基,氨基烷基,烷基,烷氧基和酮基组成; R 6是连接体; AA是氨基酸; n为1〜6; 或其盐。 还公开了药物组合物,抑制SH2结构域与磷蛋白结合的方法和治疗乳腺癌的方法。
摘要:
In a coding procedure, a spectral content of a speech signal is estimated. A preferential coding algorithm or preferential value of at least one coding parameter is selected based on the estimated spectral content of the speech signal. The speech signal is coded in accordance with the selected coding algorithm or the selected coding parameter to control the operation of one or more of the following: a pre-processing filter, a post-processing filter, a coding control coefficient, a weighting filter, a synthesis filter, and a quantization table.
摘要:
An approach for efficiently reducing background noise from speech signal in real-time applications is presented. A noisy input speech signal is processed through an inverse filter when the spectrum tilt of the input signal is not that of a pure background noise model the noisy input signal is also filtered in order to reduce the spectrum valley areas of the noisy input signal when the background noise is present.
摘要:
A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.
摘要:
A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The speech encoder also utilizes an adaptive weighting factor in the selection of a current pitch lag value from a plurality of pitch lag candidates. For example, if the speech encoder identifies an integer multiple timing relationship between any two pitch lag candidates, the pitch lag candidate with the smallest timing value is favored through adjustment of the weighting factor. Similarly, if a pitch lag candidate exhibits timing that corresponds to that of previous pitch lag values, the weighting factor is adjusted to favor that candidate.
摘要:
A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. The encoder utilizes gain normalization wherein LPC (linear predictive coding) gain provides a smoothing factor for combining both open and closed loop gains. The lower the LPC gain, the greater the open loop gain contribution to a gain normalization factor. The greater the LPC gain, the greater the closed loop gain contribution. For background noise, the smaller of the closed and open loop gains are used as the normalization factor. The normalization factor is limited by the LPC gain to prevent influencing the coding quality.