Abstract:
In a speech encoder/decoder (200/300) a pitch delay contour endpoint modifier (208) is employed to shift the endpoints of a pitch delay interpolation curve up or down. Particularly, the endpoints of the pitch delay interpolation curve are shifted based on a variation and/or a standard deviation in pitch delay.
Abstract:
A CELP encoder is provided that optimizes excitation vector-related parameters in a more efficient manner than the encoders of the prior art. In one embodiment, a CELP encoder (400) optimizes excitation vector-related parameters ( τ , β , κ , and γ ) based on a computed correlation matrix (Φ'), which matrix is in turn based on a filtered first excitation vector ( yτ ( n )). The encoder then evaluates error minimization criteria based on at least in part on a target signal ( xw ( n )), which target signal is based on an input signal ( s(n) ), and the correlation matrix and generates a excitation vector-related index in response to the error minimization criteria. In another embodiment, a CELP encoder (600) is provided that is capable of jointly optimizing and/or sequentially optimizing multiple excitation vector-related parameters by reference to a joint search weighting factor ( λ ), thereby invoking an optimal error minimization process.
Abstract translation:提供了一种CELP编码器,其以比现有技术的编码器更有效的方式优化激励矢量相关参数。 在一个实施例中,CELP编码器(400)基于所计算的相关矩阵(PHI')来优化激励矢量相关参数(tau,β,κ和γ),所述相关矩阵依次基于滤波的第一激励矢量 y tau(n))。 然后,编码器至少部分地基于目标信号(xw(n)),基于输入信号(s(n))和相关矩阵来产生误差最小化准则,并产生激励矢量 - 相关索引响应错误最小化标准。 在另一个实施例中,提供了一种CELP编码器(600),其能够通过参考联合搜索加权因子(lambda)联合优化和/或顺序优化多个与激励矢量相关的参数,由此调用最佳误差最小化处理。 >
Abstract:
Apparatus for encoding at least one parameter associated with a signal source for transmission over k frames to a decoder comprises a processor which is configured in operation to assign a predetermined bit pattern to n bits associated with the at least one parameter of a first frame of k frames and set the n bits associated with the at least one parameter of each of k-1 subsequent frames to values, such that the values of the n bits of the k-1 subsequent frames represent the at least one parameter. The predetermined bit pattern indicates a start of the at least one parameter.
Abstract:
In a selective signal encoder, an input signal is first encoded (1004) using a core layer encoder to produce a core layer encoded signal. The core layer encoded signal is decoded (1006) to produce a reconstructed signal and an error signal is generated (1008) as the difference between the reconstructed signal and the input signal. The reconstructed signal is compared (1010) to the input signal. One of two or more enhancement layer encoders selected (1014, 1016) dependent upon the comparison and used to encode the error signal. The core layer encoded signal, the enhancement layer encoded signal and the selection indicator are output (1018) to the channel (for transmission or storage, for example).
Abstract:
A communication device (10) can include a transceiver (12), a transducer or microphone (14) coupled to the transceiver for receiving voice input from a user, and a processor (16) coupled to the transceiver. The processor can be programmed to receive (32) a predetermined type of phone call or an indication of a caller indication from a calling party and warp (38) a voice input from the user of the communication device based on the caller indication. How the predetermined type of phone call is determined can be done in a number of ways including using an optional caller identification module (18). The caller ID module can provide information on whether the caller is recognized or whether the caller ID information is unavailable.
Abstract:
To achieve high quality speech reconstruction at low bit rates, constraints on position combinations among two or more pulses (403) are implemented. By placing constraints on position combinations, certain combinations of pulses are prohibited which allows the most significant pulses to always be coded, thereby improving speech quality. After all valid combinations are considered, a list of pulse pairs (codebook) which can be indexed using a single, predetermined bit length codeword is produced. The codeword is transmitted to a destination where it is used by a decoder to reconstruct the original information signal.
Abstract:
A mobile station (106) in a code division multiple access (CDMA) communication system (100) implements scanning of pilot channels on a different frequency by reserving a periodic frame on the forward channel to allow the mobile station (106) to change frequencies and scan for other pilot channels. To preserve some aspects of voice quality, both the base station (102) and mobile station (106) voice encode speech at a maximum of half rate and transmit the information as secondary traffic prior to the frame where the mobile station (106) scans the alternate frequency. To maximize the trade-off between voice quality and frequency of the scan, the base station (102) indicates to the mobile station (106) the period between other frequency scans via messaging. To ensure compatibility, this method can be negotiated via known service configuration negotiation techniques.
Abstract:
A set of peaks in a reconstructed audio vector Ŝ of a received audio signal is detected and a scaling mask Ψ (Ŝ) based on the detected set of peaks is generated. A gain vector g* is generated based on at least the scaling mask and an index j representative of the gain vector. The reconstructed audio signal is scaled with the gain vector to produce a scaled reconstructed audio signal. A distortion is generated based on the audio signal and the scaled reconstructed audio signal. The index of the gain vector based on the generated distortion is output.
Abstract:
A noise suppression system (109) implemented in communication system (700) provides an improved update decision during instances of sudden increase in background noise level. The noise suppression system (109), inter alia, generates an update by continually monitoring the deviation of spectral energy and forcing an update based on a predetermined threshold criterion. The spectral energy deviation is determined by utilizing an element which has the past values of the power spectral components exponentially weighted. The exponential weighting is a function of the current input energy, which means the higher the input signal energy, the longer the exponential window. Conversely, the lower the signal energy, the shorter the exponential window. The noise suppression system (109) also inhibits a forced update during periods of continuous, non-stationary input signals (such as "music-on-hold").
Abstract:
A communication device (10) can include a transceiver (12), a transducer or microphone (14) coupled to the transceiver for receiving voice input from a user, and a processor (16) coupled to the transceiver. The processor can be programmed to receive (32) a predetermined type of phone call or an indication of a caller indication from a calling party and warp (38) a voice input from the user of the communication device based on the caller indication. How the predetermined type of phone call is determined can be done in a number of ways including using an optional caller identification module (18). The caller ID module can provide information on whether the caller is recognized or whether the caller ID information is unavailable.