摘要:
Provided is a residual signal coding/decoding apparatus and method. The residual signal coding apparatus includes a transformer, a band splitter, a pulse searcher, and a pulse quantizer. The transformer transforms time-domain residual signals into a frequency domain to output transform coefficients. The band splitter splits the transform coefficients into bands to output the transform coefficients. The pulse searcher searches the transform coefficients for the respective bands to select optimal pulses and output parameters of the optimal pulses. The pulse quantizer quantizes the parameters of the optimal pulses.
摘要:
A post-filtering apparatus and method for speech enhancement in a modified discrete cosine transform (MDCT) domain are disclosed. In the apparatus and method, previous and current MDCT coefficients are used for obtaining a speech spectrum coefficient similar to a real speech spectrum, and a convex function is used for transforming the speech spectrum coefficient and obtaining a post-filter coefficient so that difference can increase in the case where the speech spectrum coefficient is small but decrease in the case where the coefficient is large. Then, the post-filter coefficient is applied to the MDCT coefficient. With this configuration, both the current and previous MDCT values are used, so that it is possible to obtain a spectrum coefficient similar to the real speech spectrum and to obtain a more accurate filter coefficient. Further, the coefficient is adaptively transformed through the convex function, thereby enhancing speech quality.
摘要:
An encoding apparatus is provided. The encoding apparatus includes a track structure determiner determining a track structure using frequency coefficients, a frequency coefficient allocator allocating the frequency coefficients to each track according to the determined track structure, and a quantizer quantizing one or more pulses in each track based on a number of frequency coefficients allocated to a corresponding track. The encoding apparatus can prevent the degradation of sound quality by avoiding the problem faced by most sinusoidal quantization techniques using a fixed track structure, i.e., a failure to quantize all pulses due to mismatches between the pulse distribution of frequency coefficients and a track structure.
摘要:
An encoding apparatus and a decoding apparatus for reducing the quantization error of a G.711 codec and improving sound quality are provided. The encoding apparatus includes a G.711 encoder which generates a G.711 bitstream by encoding an input audio signal; an enhancement-layer encoder which chooses one of a static bit allocation method and a dynamic bit allocation method that can produce less quantization error based on the input audio signal and the G.711 bitstream, and outputs an enhancement-layer bitstream including encoded additional mantissa information obtained by using the chosen bit allocation method; and a multiplexer which multiplexes the G.711 bitstream and the enhancement-layer bitstream. Therefore, it is possible to reduce the quantization error of a G.711 codec and improve sound quality.
摘要:
The present invention provides a pitch conversion method for reducing complexity of a transcoder for optimizing a speech quality and a complexity using characteristics of encoder in a transmitter and decoder in a receiver. The pitch conversion method for reducing complexity of the transcoder includes: classifying plural frames transmitted from a transmitter into frame units, each having a predetermined number of frame; recognizing a transmitting pitch included in the frame units; deciding a pitch estimation range based on the transmitting pitch; estimating at least one candidate pitch in the pitch estimation range by using a open-loop pitch search operation; and searching a final pitch around the estimated candidate pitch by using a closed-loop pitch search operation.
摘要:
Provided are a fixed codebook search method based on iteration-free global pulse replacement in a speech codec, and a Code-Excited Linear-Prediction (CELP)-based speech codec using the method. The fixed codebook search method based on iteration-free global pulse replacement in a speech codec includes the steps of: (a) determining an initial codevector using a pulse-position likelihood vector or a correlation vector; (b) calculating a fixed-codebook search criterion value for the initial codevector; (c) calculating fixed-codebook search criterion values for respective codevectors obtained by replacing a pulse of the initial codevector each time for respective tracks, and determining a pulse position generating the largest fixed-codebook search criterion value as a candidate pulse position for the respective tracks, respectively; (d) calculating fixed-codebook search criterion values for respective codevectors of all combinations obtained by replacing at least one pulse position of the initial codevector with the candidate pulse positions of the respective tracks, and determining the largest value of the fixed-codebook search criterion values; and (e) comparing the fixed-codebook search criterion value for the initial codevector obtained in step (b) with the largest value determined in step (d) to determine an optimum fixed codevector.
摘要:
Provided are a fixed codebook search method based on iteration-free global pulse replacement in a speech codec, and a Code-Excited Linear-Prediction (CELP)-based speech codec using the method. The fixed codebook search method based on iteration-free global pulse replacement in a speech codec includes the steps of: (a) determining an initial codevector using a pulse-position likelihood vector or a correlation vector; (b) calculating a fixed-codebook search criterion value for the initial codevector; (c) calculating fixed-codebook search criterion values for respective codevectors obtained by replacing a pulse of the initial codevector each time for respective tracks, and determining a pulse position generating the largest fixed-codebook search criterion value as a candidate pulse position for the respective tracks, respectively; (d) calculating fixed-codebook search criterion values for respective codevectors of all combinations obtained by replacing at least one pulse position of the initial codevector with the candidate pulse positions of the respective tracks, and determining the largest value of the fixed-codebook search criterion values; and (e) comparing the fixed-codebook search criterion value for the initial codevector obtained in step (b) with the largest value determined in step (d) to determine an optimum fixed codevector.
摘要:
Provided is a residual signal coding/decoding apparatus and method. The residual signal coding apparatus includes a transformer, an LPC coefficient extractor, an LPC coefficient quantizer, an LP analysis filter, a band splitter, a pulse searcher, and a pulse quantizer. The transformer transforms time-domain residual signals into a frequency domain to output transform coefficients. The LPC coefficient extractor extracts LPC coefficients from the transform coefficients. The LPC coefficient quantizer quantizes the LPC coefficients to output quantized LPC coefficients and corresponding indices. The LP analysis filter performs an LP analysis on the transform coefficients to output LP residual transform coefficients. The band splitter splits the LP residual transform coefficients into bands to output the LP residual transform coefficients. The pulse searcher searches the LP residual transform coefficients for the respective bands to select optimal pulses and output parameters of the optimal pulses. The pulse quantizer quantizes the parameters of the optimal pulses.
摘要:
A post-filtering apparatus and method for speech enhancement in a modified discrete cosine transform (MDCT) domain are disclosed. In the apparatus and method, previous and current MDCT coefficients are used for obtaining a speech spectrum coefficient similar to a real speech spectrum, and a convex function is used for transforming the speech spectrum coefficient and obtaining a post-filter coefficient so that difference can increase in the case where the speech spectrum coefficient is small but decrease in the case where the coefficient is large. Then, the post-filter coefficient is applied to the MDCT coefficient. With this configuration, both the current and previous MDCT values are used, so that it is possible to obtain a spectrum coefficient similar to the real speech spectrum and to obtain a more accurate filter coefficient. Further, the coefficient is adaptively transformed through the convex function, thereby enhancing speech quality.
摘要:
Provided is a residual signal coding/decoding apparatus and method. The residual signal coding apparatus includes a transformer, an LPC coefficient extractor, an LPC coefficient quantizer, an LP analysis filter, a band splitter, a pulse searcher, and a pulse quantizer. The transformer transforms time-domain residual signals into a frequency domain to output transform coefficients. The LPC coefficient extractor extracts LPC coefficients from the transform coefficients. The LPC coefficient quantizer quantizes the LPC coefficients to output quantized LPC coefficients and corresponding indices. The LP analysis filter performs an LP analysis on the transform coefficients to output LP residual transform coefficients. The band splitter splits the LP residual transform coefficients into bands to output the LP residual transform coefficients. The pulse searcher searches the LP residual transform coefficients for the respective bands to select optimal pulses and output parameters of the optimal pulses. The pulse quantizer quantizes the parameters of the optimal pulses.