Abstract: An apparatus and a method for controlling a vehicle may include at least two buttons provided on a steering wheel, a processor electrically connected to the at least two buttons, and storage that stores instructions executed by the processor, wherein the processor may map and set a vehicle function to the at least two buttons, detect a button manipulated among the at least two buttons, and perform a vehicle function mapped to the manipulated button based on the vehicle function mapped to the operated button and a current vehicle state.

Abstract: A method for connecting a wireless terminal to a vehicle includes checking a wired connection or a wireless connection between an infortainment system of the vehicle and the wireless terminal, recognizing identification of the wireless terminal, determining whether or not the wireless terminal is authenticated and thus suitable for connection with the infortainment system, and connecting the wireless terminal to the infortainment system only in a case in which the wireless terminal is authenticated based on a result of the determining of whether or not the wireless terminal is authenticated.

Abstract: A display apparatus includes: a first communication unit configured to receive first multimedia data and identification information of the first multimedia data through a first communication network; an address generator configured to generate access addresses of second multimedia data using the identification information; a second communication unit configured to receive the second multimedia data from the access addresses through a second communication network; an image processor configured to combine the first multimedia data and the second multimedia data so as to generate third multimedia data; and a display configured to output the third multimedia data.

Abstract: A method for processing a media signal, comprising: receiving, by an audio processing apparatus, an audio signal including a first channel signal and a second channel signal; estimating center sound by applying a band-pass filter to the first channel signal and the second channel signal; obtaining a first ambient sound by subtracting the center sound from the first channel signal; obtaining a second ambient sound by subtracting the center sound from the second channel signal; applying at least one of delay and reverberation filter to at least one of the first ambient sound and the second ambient sound to generate a processed ambient sound; and, generating pseudo surround signal using the center sound and the processed ambient sound is provided.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; receiving user gain input; generating a linear gain factor and a non-linear gain factor using the user gain input; modifying the non-linear gain factor using absolute threshold of hearing and power of the input signal to generate a modified non-linear gain factor; and, applying the linear gain factor and the modified non-linear gain factor to the audio signal.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; extracting a low frequency signal, a mid frequency signal and a high frequency signal from the input signal; obtaining at least one of a low-band gain and a harmonic control factor, based on a loudspeaker characteristic; obtaining mid-band gain based on the loudspeaker characteristic; generating a modified low frequency signal by applying the low-band gain to the low frequency signal; when the harmonic control factor is obtained, generating a harmonic signal from the modified low frequency signal using the harmonic control factor, generating a modified mid frequency signal by applying the mid-band gain to the mid frequency signal; and, generating a mixed signal by mixing the modified mid frequency signal, the high frequency signal, and at least one of the modified low frequency signal and the harmonic signal.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal.

Abstract: A method for processing an audio signal is disclosed. The present invention includes obtaining a stereophonic audio signal including a speech component signal and other component signals, obtaining gain values for each channel of the audio signal, determining whether the audio signal is an inverse-phase mono signal including left and right channel whose phase is inverted, inverting a phase of the obtained gain value corresponding to the one channel of the audio signal when the audio signal is an inverse-phase mono signal, modifying the speech component signal based on the inverted phase of the gain value, and generating a modified audio signal including the modified speech component signal, wherein the modified audio signal is in-phase mono signal.

Abstract: An apparatus for controlling a volume of an audio signal and method thereof are disclosed, by which a volume of an input signal can be controlled by using a non-linear gain curve and a target volume of the input signal. The present invention includes receiving an input signal and reference information controlling a volume of the input signal; and adjusting a volume of the input signal with the reference information, wherein the reference information comprises at least one of metadata indicating characteristics of the input signal, output environment information indicating characteristics of an environment in proximity to the apparatus and stored reference information being transmitted from memory. Therefore, the present invention is able to control a volume of an input signal adaptively by using reference information, for example metadata, outside environment information and stored reference information.

Abstract: An apparatus for controlling a volume of an audio signal and method thereof are disclosed, by which a volume of an input signal can be controlled by using a non-linear gain curve and a target volume of the input signal. The present invention includes storing a non-linear gain curve indicating relations between volumes of an input signal and an output signal; receiving an input signal and target volume information; determining a first gain by using the volume of the input signal and the target volume information; determining a second gain from one of a) the volume of the input signal and the target volume information or b) the first gain, by using the non-linear gain curve; and adjusting the volume of the input signal by applying the second gain to the input signal, wherein the non-linear gain curve is generated by using non-linearity information.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed, by which a local dynamic range of an audio signal can be adaptively normalized as well as a maximum dynamic range of the audio signal. The present invention includes receiving, by an audio processing apparatus, a signal, and feedback information estimated based on a normalizing gain; generating a noise estimation based on the signal; computing a gain filter for noise canceling, based on the noise estimation and the signal; and, obtaining a restricted gain filter by applying the feedback information to the gain filter.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed, by which a local dynamic range of an audio signal can be adaptively normalized as well as a maximum dynamic range of the audio signal. The present invention includes receiving a signal, by an audio processing apparatus; computing a long-term power and a short-term power by estimating power of the signal; generating a slow gain based on the long-term power; generating a fast gain based on the short-term power; obtaining a final gain by combining the slow gain and the fast gain; and, modifying the signal using the final gain.

Abstract: A method for processing an audio signal is disclosed. The present invention includes obtaining a plural-channel audio signal including a speech component signal and other component signals, obtaining gain of the audio signal, obtaining modification strength parameter determining the strength of the modification of the speech component signal, modifying the gain based on the modification strength parameter, and modifying the speech component signal based on the modified gain.

Abstract: The present invention relates to computing circuits and method for running an MPEG-2 AAC or MPEG-4 AAC algorithm efficiently, which is used as an audio compression algorithm in multi-channel high-quality audio systems, on programmable processors. In accordance with the present invention, the IMDCT process which takes large part of the amount of the operations in implementation of an MPEG-2/4 AAC algorithm can be performed in efficient. In addition, while the architecture of the existing digital signal processor is still used, the performance can be improved by means of the addition of the architecture of the address generator, Huffman decoder, and bit processing architecture. After all, to design and change the programmable processor is facilitated.

Abstract: A method for processing a media signal, comprising: receiving, by an audio processing apparatus, an audio signal including a first channel signal and a second channel signal; estimating center sound by applying a band-pass filter to the first channel signal and the second channel signal; obtaining a first ambient sound by subtracting the center sound from the first channel signal; obtaining a second ambient sound by subtracting the center sound from the second channel signal; applying at least one of delay and reverberation filter to at least one of the first ambient sound and the second ambient sound to generate a processed ambient sound; and, generating pseudo surround signal using the center sound and the processed ambient sound is provided.

Abstract: The present invention relates to computing circuits and method for running an MPEG-2 AAC or MPEG-4 AAC algorithm efficiently, which is used as an audio compression algorithm in multi-channel high-quality audio systems, on programmable processors. In accordance with the present invention, the IMDCT process which takes large part of the amount of the operations in implementation of an MPEG-2/4 AAC algorithm can be performed in efficient. In addition, while the architecture of the existing digital signal processor is still used, the performance can be improved by means of the addition of the architecture of the address generator, Huffman decoder, and bit processing architecture. After all, to design and change the programmable processor is facilitated.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; receiving user gain input; generating a linear gain factor and a non-linear gain factor using the user gain input; modifying the non-linear gain factor using absolute threshold of hearing and power of the input signal to generate a modified non-linear gain factor; and, applying the linear gain factor and the modified non-linear gain factor to the audio signal.

Abstract: The present invention relates to computing circuits and method for running an MPEG-2 AAC or MPEG-4 AAC algorithm efficiently, which is used as an audio compression algorithm in multi-channel high-quality audio systems, on programmable processors. In accordance with the present invention, the IMDCT process which takes large part of the amount of the operations in implementation of an MPEG-2/4 AAC algorithm can be performed in efficient. In addition, while the architecture of the existing digital signal processor is still used, the performance can be improved by means of the addition of the architecture of the address generator, Huffman decoder, and bit processing architecture. After all, to design and change the programmable processor is facilitated.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; extracting a low frequency signal, a mid frequency signal and a high frequency signal from the input signal; obtaining at least one of a low-band gain and a harmonic control factor, based on a loudspeaker characteristic; obtaining mid-band gain based on the loudspeaker characteristic; generating a modified low frequency signal by applying the low-band gain to the low frequency signal; when the harmonic control factor is obtained, generating a harmonic signal from the modified low frequency signal using the harmonic control factor, generating a modified mid frequency signal by applying the mid-band gain to the mid frequency signal; and, generating a mixed signal by mixing the modified mid frequency signal, the high frequency signal, and at least one of the modified low frequency signal and the harmonic signal.