摘要:
The method and device of the present invention provide a mechanism at a network node to compensate for variable delays or delay jitters of video packets transported over a packet-switched network, such as an ATM network, and generate a continuous bitstream to an external decoder. The received video packets with variable delays are first depacketized and stored in a video information buffer and then are sent to the external decoder through another constant or variable bit-rate channel. Based on a hypothetical decoder buffer verifier condition, stuffing bits are inserted into the output video bitstream to prevent the decoder buffer from overflowing. Stuffing bits are sent if one of the following two cases occurs: A) the condition of the hypothetical decoder buffer verifier is violated; B) the video information buffer is empty. A minimum number of stuffing bits are sent each time to minimize the incurred delay and concurrently make the following three conditions satisfied: 1) overflow of the decoder buffer is avoided; 2) idleness of the channel is avoided; 3) the stuffing bits must comply with the syntax of the video bitstream.
摘要:
A priority assignment method and device are set forth for assigning a priority to a selected speech frame coded by a linear predictive coder based on at least two of: an energy of the speech frame, a log spectral distance between a frame and a frame immediately previous, and a pitch predictor coefficient for the selected speech frame. The invention protects against loss of perceptually important and hard-to-reconstruct speech frames.
摘要:
A novel spectral interpolation and efficient excitation codebook search method developed for a Code-Excited Linear Predictive (CELP) speech coder is set forth. The interpolation is performed on an impulse response of the spectral synthesis filter. As the result of using this new set of interpolation parameters, the computations associated with an excitation codebook search in a CELP coder are considerably reduced. Furthermore, a coder utilizing this new interpolation approach provides noticeable improvement in speech quality coded at low bit-rates.