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 embodiments of the present invention relate to a compression coding and decoding method, a coder, a decoder and a coding device. The compression coding method includes: extracting sign information of an input signal to obtain an absolute value signal of the input signal; obtaining a residual signal of the absolute value signal by using a prediction coefficient, where the prediction coefficient is obtained by prediction and analysis that are performed according to a signal characteristic of the absolute value signal of the input signal; and multiplexing the residual signal, the sign information and a coding parameter to output a coding code stream, after the residual signal, the sign information and the coding parameter are respectively coded, so as to improve compression efficiency of a voice and audio signal.
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 invention relates to encoding technology. The encoding method includes selecting a second encoding mode for encoding an input frame signal according to an analysis on signal characteristic of the input frame signal; obtaining coding demand values for a preset first encoding mode and the second encoding mode which are used to encode the input frame signal; determining, from the above encoding modes based on the coding demand values, an encoding mode for encoding the input frame signal; and multiplexing information of the determined encoding mode and encoded data which are encoded according to the determined encoding mode. Hence, the compatibility and the prioritization in terms of the encoding modes can be achieved.
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:
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:
Embodiments of the present invention provide an audio signal coding and decoding method and device. The coding method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present invention, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved.
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.