Abstract: A system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, encoding a difference between a baseband speech signal and a standard baseband between a synthesized standard baseband signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

Abstract: A stereophonic sound reproduction system for compensating a low frequency signal and a method thereof, wherein a mono component signal for compensating low frequency signals which are attenuated when removing a crosstalk of inputted left and right signals inputted is calculated using an average value between the left and right signals, left and right compensation gains which are inversely proportional to an absolute value of a power difference value between the first left and right signals, an amplitude of the calculated mono component signal is controlled according to the left and right compensation gains, and thereafter the mono component signal with the controlled amplitude is added to the left and right signals when removing the crosstalk, whereby the left and right signals from which the crosstalk is removed and to which the mono component signal is added are outputted through left and right speakers to thus prevent distortion of the low frequency signals of original stereophonic sound with maintaining a

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract: The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

Abstract: The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

Abstract: A transcoder for use between speech codecs using different Code-Excited Linear Prediction (CELP) type and a method therefor are disclosed. The transcoder includes a decoding unit of an input CELP codec, a transcoding filter, a transcoding filter design unit, and an encoding unit of an output CELP codec. By substituting a post-filter and a perceptual weighting filter of a prior art with one transcoding filter, the calculation amount of the transcoder is reduced, and speech quality decoded at a receiving end is improved.

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract: An apparatus for converting voice packets transmitted/received through a network includes a first transcoder for performing at least one of bit-unpacking and unquantization on an encoded packet at a first encoder, namely transmitting party, to obtain an LSP (Line Spectrum Pair) parameter of the first encoder, and converting and unquantizing the LSP parameter to an LSP parameter of a second encoder, namely receiving party, to do bit-packing. A second transcoder performs at least one of bit-unpacking and unquantization on an encoded packet at the second encoder, namely transmitting party, to obtain an LSP parameter of the second encoder, and converts and unquantizes the LSP parameter to an LSP parameter of the first encoder, namely receiving party, to do bit-packing.

Abstract: Provided are a speech restoration system and method for concealing packet losses.

Abstract: A stereophonic sound reproduction system for compensating a low frequency signal and a method thereof, wherein a mono component signal for compensating low frequency signals which are attenuated when removing a crosstalk of inputted left and right signals inputted is calculated using an average value between the left and right signals, left and right compensation gains which are inversely proportional to an absolute value of a power difference value between the first left and right signals, an amplitude of the calculated mono component signal is controlled according to the left and right compensation gains, and thereafter the mono component signal with the controlled amplitude is added to the left and right signals when removing the crosstalk, whereby the left and right signals from which the crosstalk is removed and to which the mono component signal is added are outputted through left and right speakers to thus prevent distortion of the low frequency signals of original stereophonic sound with maintaining a

Abstract: A method for searching a codebook which predicts a residual element of an input voice signal includes combining each track of the input signal, forming track units including at least two tracks, and determining a pulse code for each track. The method further includes calculating energy for each track using an energy formula including a vector dot product, arranging or selecting codewords in a small track energy order, and searching or selecting an optimal pulse for a single- or double-pulse track of the selected codeword.

Abstract: Provided is a highband coding apparatus and method for a wideband coding system. The coding apparatus and method can reduce a pre-echo phenomenon by encoding the highband based on lowband encoding information and Temporal Noise Shaping technique. A highband encoding apparatus includes: a domain converter for converting the domain of an input highband signal into a frequency domain; a linear prediction order determiner for determining a linear prediction order based on the lowband encoding information; a linear prediction analyzer for analyzing a highband signal of the frequency domain based on the determined linear prediction order to thereby generate a linear prediction coefficient; a linear prediction coefficient quantizer for quantizing the linear prediction coefficient based on the lowband encoding information; and a residual signal quantizer for obtaining a residual signal by dequantizing the quantized linear prediction coefficient and quantizing the residual signal.

Abstract: The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract: The present invention relates to methods and devices for estimating harmonics that reduce the calculation amount and can be used very effectively in a low transmission rate voice encoder by adjusting a harmonic interval with centering on a multiple of a basic frequency or extracting a peak so that the error between an original signal spectrum and estimated harmonic spectrum is reduced.

Abstract: A method for searching a codebook which predicts a residual element of an input voice signal includes combining each track of the input signal, forming track units including at least two tracks, and determining a pulse code for each track. The method further includes calculating energy for each track using an energy formula including a vector dot product, arranging or selecting codewords in a small track energy order, and searching or selecting an optimal pulse for a single- or double-pulse track of the selected codeword.

Abstract: An apparatus for converting voice packets transmitted/received through a network includes a first transcoder for performing at least one of bit-unpacking and unquantization on an encoded packet at a first encoder, namely transmitting party, to obtain an LSP (Line Spectrum Pair) parameter of the first encoder, and converting and unquantizing the LSP parameter to an LSP parameter of a second encoder, namely receiving party, to do bit-packing. A second transcoder performs at least one of bit-unpacking and unquantization on an encoded packet at the second encoder, namely transmitting party, to obtain an LSP parameter of the second encoder, and converts and unquantizes the LSP parameter to an LSP parameter of the first encoder, namely receiving party, to do bit-packing.