摘要:
A voice packet forwarding apparatus and method is provided in a digital communication system including a switched vocoder module to directly pass a voice packet received from a packet terminal to a digital communication network or a PCM signal decoded from the voice packet to the digital communication network. The switched vocoder module is also provided to directly pass a voice packet received from the digital communication network to the packet terminal or transmit a voice packet coded from a PCM signal received from the digital communication network. In the presence of additional data to be transmitted to the packet terminal, a data inserter is provided to insert the additional data in the voice packet received from the switched vocoder module and transmit the voice packet with the additional data to the packet terminal. A controller is provided to control the switched vocoder module and the data inserter.
摘要:
A voice packet forwarding apparatus and method is provided in a digital communication system including a switched vocoder module to directly pass a voice packet received from a packet terminal to a digital communication network or a PCM signal decoded from the voice packet to the digital communication network. The switched vocoder module is also provided to directly pass a voice packet received from the digital communication network to the packet terminal or transmit a voice packet coded from a PCM signal received from the digital communication network. In the presence of additional data to be transmitted to the packet terminal, a data inserter is provided to insert the additional data in the voice packet received from the switched vocoder module and transmit the voice packet with the additional data to the packet terminal. A controller is provided to control the switched vocoder module and the data inserter.
摘要:
An apparatus and a method for transmitting audio signals in such a manner that audio signals transmitted and received in a mobile telephone network are preprocessed before being inputted to a voice encoder are provided. The audio signals are preprocessed by using an optimal filter gain based on error signals obtained when the audio signals are preprocessed and outputted by the voice encoder and the synthesizer. Therefore, the sound quality of the audio signals is hardly degraded by the voice encoder.
摘要:
Provided is a voice packet rate converting method and apparatus. The rate of at least one first element of an input voice packet compressed at a first rate is converted to a second rate at a PCM level, and the rate of a second element of the input voice packet is converted to the second rate at a parameter level. An output voice packet compressed at the second rate is generated by combining the first element and second elements at the second rate.
摘要:
An apparatus and method for recovering lost voice packets are provided, in which a packet loss detector determines whether a received packet has been lost, packet information storage stores voice information of previous voice packets and voice information of the received voice packet, a packet error corrector measures the voice information of the received voice packet, stores the measured voice information in the packet information storage, corrects the voice information when necessary, and generates a corrected voice packet, if the received voice packet is normal, and a packet loss recoverer recovers the voice information of the received voice packet using the voice information of previous voice packets stored in the packet information storage and generates a recovered voice packet, if the received voice packet has been lost.
摘要:
An apparatus and method for recovering lost voice packets are provided, in which a packet loss detector determines whether a received packet has been lost, packet information storage stores voice information of previous voice packets and voice information of the received voice packet, a packet error corrector measures the voice information of the received voice packet, stores the measured voice information in the packet information storage, corrects the voice information when necessary, and generates a corrected voice packet, if the received voice packet is normal, and a packet loss recoverer recovers the voice information of the received voice packet using the voice information of previous voice packets stored in the packet information storage and generates a recovered voice packet, if the received voice packet has been lost.
摘要:
Provided are an apparatus for multi-stage transforming a plurality of unit blocks in multi-dimension that can improve compression efficiency of video data by collecting Discrete Cosine Transforming (DCT) coefficients of neighboring blocks and performing an additional transformation based on the DCT coefficients of an original picture and a differential picture. The method includes the steps of: performing a Discrete Cosine Transform (DCT) on inputted picture data and selecting R blocks of a predetermined size from DCT picture data, where R is a natural number equal to or greater than 2; arranging DCT coefficients of each of the selected R blocks according to each frequency in one-dimension; and performing one-dimensional transformation again on the DCT coefficients arranged in one-dimension.
摘要:
Provided are an apparatus for multi-stage transforming a plurality of unit blocks in multi-dimension that can improve compression efficiency of video data by collecting Discrete Cosine Transforming (DCT) coefficients of neighboring blocks and performing an additional transformation based on the DCT coefficients of an original picture and a differential picture. The method includes the steps of: performing a Discrete Cosine Transform (DCT) on inputted picture data and selecting R blocks of a predetermined size from DCT picture data, where R is a natural number equal to or greater than 2; arranging DCT coefficients of each of the selected R blocks according to each frequency in one-dimension; and performing one-dimensional transformation again on the DCT coefficients arranged in one-dimension.
摘要:
An apparatus and method for deciding an adaptive noise level for bandwidth extension are provided. The apparatus includes a noise level decider for deciding a high-band noise level for bandwidth extension according to tonality of an input signal, a pitch frequency analyzer for detecting a pitch frequency of the input signal and analyzing correlation between the detected pitch frequency and a frequency channel, and a noise level controller for adaptively controlling the decided high-band noise level based on the analyzed correlation of the pitch frequency and the frequency channel.
摘要:
An echo canceler for canceling an echo signal mixed in an input signal received from a hybrid circuit. The echo canceler has a double filter structure comprised of an adaptive filter and a fixed filter. The adaptive filter receives the input signal to generate a first echo estimation signal according to a filter coefficient thereof, and the fixed filter receives the input signal to generate a second echo estimation signal according to a filter coefficient thereof. A first adder subtracts the first echo estimation signal from the input signal to generate a first echo-canceled signal, and a second adder subtracts the second echo estimation signal from the input signal to generate a second echo-canceled signal. A mode selector selectively outputs one of the first and second echo-canceled signals having a relatively more canceled echo component.