Abstract:
A method and apparatus of generating a sound field effect is provided. The sound field effect generating apparatus may generate a frequency coefficient that is frequency-transformed from a direct signal, may generate a reflection signal from the frequency coefficient, may generate an output signal using the frequency coefficient and the reflection signal, and may perform an inverse-frequency transform of the output signal.
Abstract:
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.
Abstract:
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.
Abstract:
Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.
Abstract:
Provided is an audio data decoding system and method that may selectively transfer compressed audio data from a top system to at least one audio input buffer of a separately provided and/or separately power management controlled subsystem, restore the compressed audio data to digital pulse code modulation (PCM) data using an audio decoding unit of the subsystem, convert the digital PCM data to analog PCM data or an audio output signal, and output the converted audio output signal.
Abstract:
A surround sound virtualization apparatus and method. The surround sound virtualization apparatus may include an audio decoder to perform head-related transfer function (HRTF) filtering, and a time delay unit to provide a time delay to a plurality of output signals of the audio decoder.
Abstract:
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.
Abstract:
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.
Abstract:
An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.
Abstract:
Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.