摘要:
A method and apparatus to perform bandwidth extension encoding and decoding encodes and/or decodes a high frequency signal using an excitation signal for a low frequency signal encoded in a time domain or a frequency domain or using an excitation spectrum for the low frequency signal. Accordingly, although an audio signal is encoded or decoded using a small number of bits, the quality of sound corresponding to a signal in a high frequency band does not degrade. Therefore, a coding efficiency of the audio signal can be maximized.
摘要:
A method and apparatus to extract an important frequency component of an audio signal and a method and apparatus to encode and/or decode an audio signal by using the same. The method of extracting an important frequency component of an audio signal includes converting an audio signal of a time domain into an audio signal of a frequency domain, selecting a frequency band having a harmonic feature from the converted audio signal of the frequency domain, and extracting an important frequency component from the selected frequency band having the harmonic feature.
摘要:
An apparatus to compress a wide-band speech signal, the apparatus including a narrow-band speech compressor to compress a low-band speech signal of the wide-band speech signal and output the compressed low-band speech signal as a low-band speech packet; and a high-band speech compressor to compress a high-band speech signal of the wide-band speech signal using energy information of the low-band speech signal provided from the narrow-band speech compressor, and outputs the compressed high-band speech signal as a high-band speech packet.
摘要:
A memory management method is provided. In the method, a spatial parameter included in an encoding result is represented as a vector of a time slot and a frequency band in a first domain, a temporary matrix is calculated in the first domain by using the difference between vectors of a current time slot and a previous time slot at the same frequency band and then is stored in a memory, and then a matrix needed to decode the encoding result is represented as a matrix for a time slot and a frequency band in a second domain by using the temporary matrix, thereby reducing the load on the memory for storing matrices on which a decoding operation is performed.
摘要:
A method and apparatus for implementing fixed codebooks as a common module are provided. In the method of implementing fixed codebooks of a plurality of speech codecs as a common module, it is possible to include only a part excluding fixed codebooks in a communication terminal or communication system, support various speech codecs without using a chip with high price and high performance, and reduce a memory space that is occupied by the speech codecs by generating a track of a fixed codebook corresponding to a speech codec based on information on the speech codec among the plurality of speech codecs and selecting a codebook vector corresponding to a target signal among codebook vectors constructed with combinations of pulses represented by the generated track. In addition, it is possible to reduce processing complexity as compared with a case of embodying the common fixed codebook module in software by embodying the common fixed codebook module in hardware. In addition, it is possible to improve the entire voice processing performance by applying the latest fixed codebook searching algorithm only to the common fixed codebook, thereby easily applying the latest fixed codebook searching algorithm to the entire voice codec.
摘要:
A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.
摘要:
A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.
摘要:
An apparatus to compress a wide-band speech signal, the apparatus including a narrow-band speech compressor to compress a low-band speech signal of the wide-band speech signal and output the compressed low-band speech signal as a low-band speech packet; and a high-band speech compressor to compress a high-band speech signal of the wide-band speech signal using energy information of the low-band speech signal provided from the narrow-band speech compressor, and outputs the compressed high-band speech signal as a high-band speech packet.
摘要:
A method and apparatus for implementing fixed codebooks as a common module are provided. In the method of implementing fixed codebooks of a plurality of speech codecs as a common module, it is possible to include only a part excluding fixed codebooks in a communication terminal or communication system, support various speech codecs without using a chip with high price and high performance, and reduce a memory space that is occupied by the speech codecs by generating a track of a fixed codebook corresponding to a speech codec based on information on the speech codec among the plurality of speech codecs and selecting a codebook vector corresponding to a target signal among codebook vectors constructed with combinations of pulses represented by the generated track. In addition, it is possible to reduce processing complexity as compared with a case of embodying the common fixed codebook module in software by embodying the common fixed codebook module in hardware. In addition, it is possible to improve the entire voice processing performance by applying the latest fixed codebook searching algorithm only to the common fixed codebook, thereby easily applying the latest fixed codebook searching algorithm to the entire voice codec.
摘要:
An apparatus and a method for computing a Speech Absence Probability (SAP), and an apparatus and a method for removing noise by using the SAP computing device and method are provided. The provided SAP computing device for computing the SAP indicating probability that speech is absent in a mth frame, from a first through Ncth posteriori (Nc means the total number of channels) Signal to Noise Ratios (SNR) calculated with regard to the mth frame of a speech signal and a first through Ncth predicted SNRs predicted with regard to the mth frame, includes: a first through Ncth likelihood ratio generators for generating a first through Ncth likelihood ratios from the first through Ncth posterior SNRs and the first through Ncth predicted SNRs, and outputting them; a first multiplying unit for multiplying the first through Ncth likelihood ratios by a predetermined a priori probability, and outputting the multiplication results; an adding unit for adding each of the multiplication results received from the first multiplying unit to a predetermined value, and outputting the added results; a second multiplying unit for multiplying the added results received from the adding unit and outputting the multiplication result; and a inverse number calculator for calculating inverse number of the multiplication result received from the second multiplying unit and outputting the calculated inverse number as the SAP. Therefore, since the accuracy of the calculated SAP is high, noise can be efficiently removed from the speech signal that may have noise and an enhanced speech signal with an enhanced quality can be provided.