摘要:
Perceived interactivity in user communications is achieved by reducing a perceived delay switching the active transmitter in the communication without having to reduce actual transmission and setup delays associated with a communication exchange. A sound signal is identified in the user communication. The sound signal is analyzed to identify or estimate a sound signal segment. The sound signal segment is preferably (though not necessarily) located at the beginning or the end of the sound signal. The sound signal segment may be selected directly from the sound signal itself, from a modified version of the sound signal, or from a signal associated with the sound signal. A determination is made that a length or duration of the sound signal segment should be or can be modified. One or more modifications for the sound signal segment are determined and are provided to one or more processing units to perform the modification(s).
摘要:
The present invention is to select an adaptation scheme for the transmission of the encoded media that results in a satisfactory performance of the transmitted encoded media. A difference from the prior art is that each adaptation scheme defines a set of different transmission formats, wherein each transmission formats is a combination of at least two of the parameters the source codec bit rate, the packet rate, the number of frames of each packet (referred to as frame aggregation), and the level of redundancy. By using the different transmission formats, the transmission can be adapted to different operating scenarios and the performance is hence improved.
摘要:
A multi-codebook fixed bitrate CELP signal block encoder/decoder includes a codebook selector (22) for selecting, for each signal block, a corresponding codebook identification in accordance with a deterministic selection procedure that is independent of signal type. Included are also means for encoding/decoding each signal block by using a codebook having the selected codebook identification.
摘要:
A determination is made of the positions within a speech frame for a given number of excitation pulses in a linear predictive speech encoder. A combination of two known methods is used. The positions of the excitation pulses are calculated in a number of calculation stages according to the first known method. The positions of the excitation pulses are then calculated in a number of calculation stages in accordance with the second method to obtain one of a number of pulse placements. Each calculation according to the second method begins at a starting point from one of a number of positions calculated in accordance with the first method. The proportion between the number of calculation stages in the first method and the second method is chosen so as to obtain the least calculation complexity for a certain given speech quality.
摘要:
Speech frames of a first speech coding scheme are utilized as speech frames of a second speech coding scheme, where the speech coding schemes use similar core compression schemes for the speech frames, preferably bit stream compatible. An occurrence of a state mismatch in an energy parameter between the first speech coding scheme and the second speech coding scheme is identified, preferably either by determining an occurrence of a predetermined speech evolution, such as a speech type transition, e.g. an onset of speech following a period of speech inactivity, or by tentative decoding of the energy parameter in the two encoding schemes followed by a comparison. Subsequently, the energy parameter in at least one frame of the second speech coding scheme following the occurrence of the state mismatch is adjusted. The present invention also presents transcoders and communications systems providing such transcoding functionality.
摘要:
A multi-channel linear predictive analysis-by-synthesis signal encoding method detects (S26, S27) inter-channel correlation and select one of several possible encoding modes (S24, S29, S30) based on the detected correlation.
摘要:
Interoperability is achieved between wireless user communication devices that have different speech processing formats and/or attributes. A first wireless user communication device includes a primary speech codec that encodes a first speech message using a first speech encoding format. The encoded speech is then sent to a second wireless user communications device that includes a primary speech codec supporting a second speech encoding format. The first user device receives from the second user device a second speech message encoded using the second speech encoding format. The second speech message is then decoded by the first user device using a second speech decoder supporting decoding of the second speech encoding format. But the first communication device does not support speech encoding using the second speech encoding format—regardless of whether the first communication device includes or does not includes an encoder for encoding speech using the first speech encoding format.
摘要:
A vehicle door is provided with a reinforcing structure which improves handling of impact energy in the longitudinal direction of the vehicle by transferring impact energy received by the A-pillar to the door through the door hinges and the reinforcing structure to the B-pillar of the vehicle frame.
摘要:
The present invention relates to speech coding in wireless and wireline communication systems. The present invention provides a method of saving bandwidth by a controlled dropping of speech frames at an encoder in a sending communication device. The dropping is controlled in a manner to minimize the effects on the speech quality after the decoding in the receiving communication device, by assuring that the state mismatch between the encoder and the decoder is removed or at least significantly reduced. This is achieved by letting the encoder run an ECU algorithm with a similar behavior as the one running in the decoder in the receiving communication device.
摘要:
The present invention is to select an adaptation scheme for the transmission of the encoded media that results in a satisfactory performance of the transmitted encoded media. A difference from the prior art is that each adaptation scheme defines a set of different transmission formats, wherein each transmission formats is a combination of at least two of the parameters the source codec bit rate, the packet rate, the number of frames of each packet (referred to as frame aggregation), and the level of redundancy. By using the different transmission formats, the transmission can be adapted to different operating scenarios and the performance is hence improved.