Abstract: A multimedia terminal comprising: a first encoder (100) for producing a first bit-stream (107) of a first media type and having a first bit-rate; a second encoder (110) for producing a second bit-stream (112) of a second media type and having a second bit-rate; a multiplexer (120) for combining at least the first (106) and the second (112) bit-streams into a third bit-stream (123). The terminal comprises an input element (130) for receiving preference information (131) coupled to the first encoder (100) and the second encoder (110), said preference information (131) indicating a preferred combination of the first and the second media types in the third bit-stream and affecting the first and the second bit-rates. Thus, the transmission capacity is utilised in a more optimised manner and the proportions of different media types are better adjusted to the purpose of the information transfer.

Abstract: A scalable encoder (100) for encoding a media signal is described. It comprises first encoding means (210) for producing a first data stream (102), which is a core data stream relating to the media signal (101), having a first bit-rate; second encoding means (230) for producing a second data stream (103), which comprises a set of enhancement data streams relating to the media signal, having a second bit-rate; and a multiplexer (110) for combining at least the first data stream and the second data stream into a third data stream (104). The scalable encoder it further comprises control means (420, 421, 422), which is arranged to receive control information (401), to determine a target combination of the first data stream and the second data stream in the third data stream according to the control information and to adjust the combination of the first data stream and the second data stream in the third data stream by affecting the first and the second bit-rates.

Abstract: A speech coding method and device for encoding and decoding an input signal and providing synthesized speech, wherein the higher frequency components of the synthesized speech are achieved by high-pass filtering and coloring an artificial signal to provide a processed artificial signal. The processed artificial signal is scaled by a first scaling factor during the active speech periods of the input signal and a second scaling factor during the non-active speech periods, wherein the first scaling factor is characteristic of the higher frequency band of the input signal and the second scaling factor is characteristic of the lower frequency band of the input signal. In particular, the second scaling factor is estimated based on the lower frequency components of the synthesized speech and the coloring of the artificial signal is based on the linear predictive coding coefficients characteristic of the lower frequency of the input signal.

Abstract: The invention relates to the coding of speech at a variable bit rate, whereby the bit rates can vary from frame to frame, and more specifically to the methods and filters used for improving the quality of the decoded speech. In the solution according to the invention the weighting factors of the postfilter are not adapted on the basis of the momentary bit rate or the bit rate used in the coding of each frame, but the weighting factors are adapted according to the average bit rate calculated on the basis of a predetermined length of time. In addition to this, the weighting factors of the postfilter are also adjusted on the basis of whether the frame in question contains a voiced speech signal, an unvoiced speech signal or background noise. At frames containing an unvoiced speech signal or background noise, postfiltering is weakened so as to avoid the distortion of the signal tone because the postfiltering is adapted to a voiced signal.

Abstract: Circuitry, embodied in a media subsystem (10A), reproduces a speech or other type of audio signal, and is operable when playing back audio data for reducing synchronization delay. A method operates by, when a frame containing audio data is sent to a decoder (20), measuring the synchronization delay; determining by how much the synchronization delay should be adjusted; and adjusting the synchronization delay in a content-aware manner by adding or removing one or more audio samples in a selected current frame or in a selected subsequent frame so as not to significantly degrade the quality of the played back audio data. When the synchronization delay is adjusted by more than one audio sample, the adjustment can be made by all of the determined audio samples in one adjustment, or the adjustment is made by less than all of the determined audio samples by a plurality of adjustments. The step of adjusting selects, if possible, an unvoiced frame and discriminates against a transient frame.

Abstract: A variable bit-rate speech coding method determines for each subframe a quantised vector d(i) comprising a variable number of pulses. An excitation vector c(i) for exciting LTP and LPC synthesis filters is derived by filtering the quantised vector d(i), and a gain value gc is determined for scaling the pulse amplitude excitation vector c(i) such that the scaled excitation vector represents the weighted residual signal {tilde over (s)} remaining in the subframe speech signal after removal of redundant information by LPC and LTP analysis. A predicted gain value ĝc is determined from previously processed subframes, and as a function of the energy Ec contained in the excitation vector c(i) when the amplitude of that vector is scaled in dependence upon the number of pulses m in the quantised vector d(i). A quantised gain correction factor {circumflex over (&ggr;)}gc is then determined using the gain value gc and the predicted gain value ĝc.

Abstract: A method of coding a sampled speech signal in which the speech signal is divided into sequential frames. For each current frame, a first set of linear prediction coding (LPC) coefficients are generated, where the number of LPC coefficients depends upon the characteristics of the current frame. If the number of LPC coefficients in the first set of the current frame differs from the number in the first set of the preceding frame, then a second expanded or contracted set of LPC coefficients is generated from the first set of LPC coefficients for the preceding frame. This second set contains the same number of LPC coefficients as are present in said first set of the current frame. Respective sets of line spectral frequency (LSP) coefficients are generated for the first set of LPC coefficients of the current frame and the second set of LPC coefficients of the preceding frame. The sets of LSP coefficients are then combined to provide an encoded residual signal.

Abstract: A method of speech coding a sampled speech signal using long term prediction (LTP). A LTP pitch-lag parameter is determined for each frame of the speech signal by first determining the autocorrelation function for the frame within the signal, between predefined maximum and minimum delays. The autocorrelation function is then weighted to emphasize the function for delays in the neighborhood of the pitch-lag parameter determined for the most recent voiced frame. The maximum value for the weighted autocorrelation function is then found and identified as the pitch-lag parameter for the frame.

Abstract: The invention is related digital speech encoding. In a speech codec according to the invention, for modeling a speech signal (301) both prediction parameters (321, 322, 331) modeling a speech signal in a short term and prediction parameters (341, 342, 351) modeling a speech signal in a long term are used. Each prediction parameter (321, 322, 331, 341, 342, 351) is presented using a certain accuracy, in a digital system with a certain number of bits. In speech encoding according to the invention the number of bits used for presenting prediction parameters (321, 322, 331, 341, 342, 351) is adjusted based upon information parameters (321, 322, 331, 341, 342, 351) obtained from a short-term LPC-analysis (32) and from a long-term LTP-analysis (31, 34, 35). The invention is particularly suitable for use at low data transfer speeds, because it offers a speech encoding method of even quality and low average bit rate.