摘要:
This invention proposes a more efficient way to quantize temporal envelope shaping of high band signal by benefiting from energy relationship between low band signal and high band signal; if low band signal is well coded or it is coded with time domain codec such as CELP, temporal envelope shaping information of low band signal can be used to predict temporal envelope shaping of high band signal; the temporal envelope shaping prediction can bring significant saving of bits to precisely quantize temporal envelope shaping of high band signal. This prediction approach can be combined with other specific approach to further increase the efficiency and save mores bits.
摘要:
A speech coding method of significantly reducing error propagation due to voice packet loss, while still greatly profiting from a pitch prediction or Long-Term Prediction (LTP), is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class; a pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. Speech coding quality loss due to the pitch gain reduction is compensated by increasing a bit rate of a second excitation component or adding one more stage of excitation component only for the first subframe or the first two subframes within the speech frame.
摘要:
Method of forming a Hybrid Split Gate Semiconductor. In accordance with a method embodiment of the present invention, a plurality of first trenches is formed in a semiconductor substrate to a first depth. A plurality of second trenches is formed in the semiconductor substrate to a second depth. The first plurality of trenches are parallel with the second plurality of trenches. The trenches of the plurality of first trenches alternate with and are adjacent to trenches of the plurality of second trenches.
摘要:
In an embodiment in accordance with the present invention, a semiconductor device includes a vertical channel region, a gate at a first depth on a first side of the vertical channel region, a shield electrode at a second depth on the first side of the vertical channel region, and a hybrid gate at the first depth on a second side of the vertical channel region. The region below the hybrid gate on the second side of the vertical channel region is free of any electrodes.
摘要:
A method of speech encoding comprises generating a first synthesized speech signal from a first excitation signal, weighting the first synthesized speech signal using a first error weighting filter to generate a first weighted speech signal, generating a second synthesized speech signal from a second excitation signal, weighting the second synthesized speech signal using a second error weighting filter to generate a second weighted speech signal, and generating an error signal using the first weighted speech signal and the second weighted speech signal, wherein the first error weighting filter is different from the second error weighting filter. The method may further generate the error signal by weighting the speech signal using a third error weighting filter to generate a third weighted speech signal, and subtracting the first weighted speech signal and the second weighted speech signal from the third weighted speech signal to generate the error signal.
摘要:
The present invention discloses a method and system for media modification, which is used for solving the problem that it is not considered during media modification whether the user confirmation or precondition of media modification is required, thereby resulting in technical defects, such as possible bearing resource embezzlement, producing redundancy, etc. The present invention regards the 200OK message of Re-INVITE as the confirmed time point of the media addition and/or modification; in the case that there is a precondition and the user confirmation is not required, the time point when the precondition is satisfied is regarded as the confirmed time point of the media component addition and/or existing media component modification; in the case that there is a precondition and the user confirmation is required, the 200OK response message of the Re-INVITE message is regarded as the confirmed time point of the media component addition and/or existing media component modification.
摘要:
The present disclosure provides a method and an Application Server (AS) for obtaining user's capability in Third Party Call Control (3PCC), and the method comprises: when a user performs 3PCC, an AS stores the media information of the user; the AS performs media negotiation with the user according to the stored media information of the user. The application of the present disclosure solves the problem of the high failure probability of media negotiation caused by the current incorrect method for obtaining the user's media capability, and improves the utilization experience of users.
摘要:
There is provided a method for use by a speech encoder to encode an input speech signal. The method comprises receiving the input speech signal; determining whether the input speech signal includes an active speech signal or an inactive speech signal; low-pass filtering the inactive speech signal to generate a narrowband inactive speech signal; high-pass filtering the inactive speech signal to generate a high-band inactive speech signal; encoding the narrowband inactive speech signal using a narrowband inactive speech encoder to generate an encoded narrowband inactive speech; generating a low-to-high auxiliary signal by the narrowband inactive speech encoder based on the narrowband inactive speech signal; encoding the high-band inactive speech signal using a wideband inactive speech encoder to generate an encoded wideband inactive speech based on the low-to-high auxiliary signal from the narrowband inactive speech encoder; and transmitting the encoded narrowband inactive speech and the encoded wideband inactive speech.
摘要:
There is provided a method of post-processing a speech signal. The method comprises applying a time-domain post-processing to the speech signal, using LPC coefficients, for a low-band frequency range and applying a frequency-domain post-processing to the speech signal, using MDCT coefficients, for the high-band frequency range. Applying the frequency-domain post-processing includes decoding an encoded speech signal to obtain MDCT coefficients representative of the speech signal divided into a plurality of sub-bands, generating an envelope for each sub-band of the plurality of sub-bands as an average magnitude of the MDCT coefficients of the sub-band, generating an envelope modification factor for each sub-band of the plurality of sub-band using the MDCT coefficients of the sub-band, modifying the envelope by the envelope modification factor for each sub-band of the plurality of sub-bands to provide a modified envelope, and generating the post-processed speech signal using the modified envelope.
摘要:
A method of performing BandWidth Extension (BWE) includes a frequency band shifting approach to generate an extended high band signal in time domain and a gain determination approach of controlling the energy of the extended high band. The proposed approach allows shifting any size of low band to any size of high band. The BWE scaling gain is estimated by using available filter bank coefficients with extremely low bit rate or without costing any bit, combining three possible gain factors.