摘要:
A frame compensation method is provided. The method includes: obtaining a length of a lost frame and a length of a correct frame; determining that the length of the correct frame is integral power of 2 times of the length of the lost frame, and then obtaining a data sequence with the same length as the length of the lost frame according to the correct frame; and compensating the lost frame according to the data sequence to obtain a compensated data frame. A frame compensation system is also provided. Lost frames in various formats are compensated according to correct frames in various formats, so that the limitation of the related art that a lost frame in a single format can be merely compensated according to a correct frame in a single format is eliminated, and the effect of the compensated data frames is better than that of filling comfort noises.
摘要:
The present invention discloses a method for obtaining an attenuation factor. The method is adapted to process the synthesized signal in packet loss concealment, and includes: obtaining a change trend of a signal; obtaining an attenuation factor, according to the change trend of the signal. The present invention also discloses an apparatus for obtaining an attenuation factor. A self-adaptive attenuation factor is adjusted dynamically by using the latest change trend of a history signal by using the present invention. The smooth transition from the history data to the data last received is realized so that the attenuation speed is kept consistent between the compensated signal and the original signal as much as possible for adapting to the characteristic of various human voices.
摘要:
The present invention discloses a method for obtaining an attenuation factor. The method is adapted to process the synthesized signal in packet loss concealment, and includes: obtaining a change trend of a pitch of a signal; obtaining an attenuation factor, according to the change trend of the pitch of the signal. The present invention also discloses an apparatus for obtaining an attenuation factor. A self-adaptive attenuation factor is adjusted dynamically by using the latest change trend of a history signal by using the present invention. The smooth transition from the history data to the data last received is realized so that the attenuation speed is kept consistent between the compensated signal and the original signal as much as possible for adapting to the characteristic of various human voices.
摘要:
The present invention discloses a signal processing method adapted to process a synthesized signal in packet loss concealment. The method includes the following steps: receiving a good frame following a lost frame, obtaining an energy ratio of energy of a signal in the signal of the good frame signal to energy of a synthesized signal corresponding to the same time of the good frame; and adjusting the synthesized signal in accordance with the energy ratio. The present invention also discloses a signal processing apparatus and a voice decoder. Through using the method provided by the present invention, the synthesized signal is adjusted in accordance with the energy ratio of the energy of the first good frame following the lost frame to the energy of the synthesized signal to ensure that there be not a waveform sudden change or an energy sudden change at the place where the lost frame and the first good frame following the lost frame are jointed in the synthesized signal, to realize the waveform's smooth transition and to avoid music noises.
摘要:
The present invention discloses a signal processing method adapted to process a synthesized signal in packet loss concealment. The method includes the following steps: receiving a good frame following a lost frame, obtaining an energy ratio of energy of a signal in the signal of the good frame signal to energy of a synthesized signal corresponding to the same time of the good frame, and adjusting the synthesized signal in accordance with the energy ratio. The present invention also discloses a signal processing apparatus and a voice decoder. Through using the method provided by the present invention, the synthesized signal is adjusted in accordance with the energy ratio of the energy of the first good frame following the lost frame to the energy of the synthesized signal to ensure that there be not a waveform sudden change or an energy sudden change at the place where the lost frame and the first good frame following the lost frame are jointed in the synthesized signal, to realize the waveform's smooth transition and to avoid music noises.
摘要:
A decoding method and device are provided. The spectrum parameter of a current bad data frame is determined. Specifically, a number of continuous bad frames that occur currently is determined. A spectrum parameter of a good data frame before the current bad data frame is determined. And a constant mean value of a spectrum parameter is determined. Then, the spectrum parameter of the good data frame is adaptively shifted towards the constant mean value of the spectrum parameter according to the number of the continuous bad data frames to calculate and obtain spectrum parameter information of the current bad frame. When the continuous bad data frames occur, the relevance between the spectrum parameter of the nearest good frame and the spectrum parameter of the current bad frame is gradually reduced, so that more accurate spectrum parameter of the current bad data frame can be obtained, thereby obtaining a better speech quality under a same code rate and a same frame error rate.
摘要:
The present disclosure relates to a method, apparatus, and system for encoding and decoding signals. The encoding method includes: converting a first-domain signal into a second-domain signal; performing Linear Prediction (LP) processing and Long-Term Prediction (LTP) processing for the second-domain signal; obtaining a long-term flag according to decision criteria; obtaining a second-domain contribution signal according to the LP processing result and the LTP processing result when the long-term flag is a first flag; obtaining a second-domain contribution signal according to the LP processing result when the long-term flag is a second flag; converting the second-domain contribution signal into a first-domain contribution signal, and calculating a first-domain predictive residual signal; and outputting a bit stream that includes the first-domain predictive residual signal. With the present disclosure, a subsequent encoding or decoding process is performed adaptively according to the long-term flag; when the long-term flag is the second flag, it is not necessary to consider the LTP processing result, thus improving the compression performance of a codec.
摘要:
The present disclosure relates to coding and decoding technologies, and discloses a preprocessing method, a preprocessing apparatus, and a coding device. The preprocessing method includes: obtaining characteristic information of a current frame signal; identifying whether the current frame signal requires no coding operation of removing LTC according to the characteristic information of the current frame signal and preset information; and if identifying that the current frame signal requires no coding operation of removing LTC, performing the coding operation of removing STC for the current frame signal; and if identifying that the current frame signal requires the coding operation of removing LTC, performing the coding operations of removing both LTC and STC for the current frame signal. Through the technical solution provided herein, the coding operation of removing LTC is performed for only part of the input frame signals.
摘要:
A method and an apparatus for processing a signal are provided. The method includes: obtaining an energy average value of each sub-band for a current frame frequency-domain signal; obtaining a current frame modification coefficient of each sub-band for the current frame frequency-domain signal according to a spectral envelope and the energy average value of each sub-band; obtaining a weighted modification coefficient of each sub-band for the current frame frequency-domain signal by using the current frame modification coefficient and a relevant frame modification coefficient; and modifying the spectral envelope of each sub-band for the current frame frequency-domain signal by using the weighted modification coefficient.
摘要:
In the field of audio encoding/decoding technologies, a signal de-noising method is provided. The method includes: selecting, according to a degree of inter-frame correlation of a frame where a spectral coefficient to be adjusted resides, at least two spectral coefficients having high correlation with the spectral coefficient to be adjusted; performing weighting on the at least two selected spectral coefficients and the spectral coefficient to be adjusted to acquire a predicted value of the spectral coefficient to be adjusted; and adjusting a spectrum of a decoded signal by using the acquired predicted value, and outputting the adjusted decoded signal. A signal de-noising apparatus corresponding to the signal de-noising method and an audio decoding system using the signal de-noising apparatus are also provided.