Abstract: A method including determining location of at least one second device relative to a first device, where at least two of the devices are configured to play audio sounds based upon audio signals; and mixing at least two of the audio signals based, at least partially, upon the determined location(s).

Abstract: A method including determining location of at least one second device relative to a first device, where at least two of the devices are configured to play audio sounds based upon audio signals; and mixing at least two of the audio signals based, at least partially, upon the determined location(s).

Abstract: A method including determining location of at least one second device relative to a first device, where at least two of the devices are configured to play audio sounds based upon audio signals; and mixing at least two of the audio signals based, at least partially, upon the determined location(s).

Abstract: A method including determining location of at least one second device relative to a first device, where at least two of the devices are configured to play audio sounds based upon audio signals; and mixing at least two of the audio signals based, at least partially, upon the determined location(s).

Abstract: Prior to a call, a command vocabulary (13) and control signals (14) corresponding to the command words are stored in the terminal. A phone number is dialed (21) and the call is connected. Speech from the user is received (24) in the terminal. A command word in the user's speech (26) is recognized (25) in the terminal. A control signal corresponding to the command word is read (27) in the terminal, and the control signal read is sent (28) from the terminal to the system in order to control the telecommunication service. The control signal is formed e.g. of a dual tone multifrequency (DTMF) code sequence.

Abstract: An improved method converts a first sampling rate of signal values to a second sampling rate, the second sampling rate being a non-integer multiple of the first sampling rate. The method includes the steps of: interpolating between successive signal values occurring at the first sampling rate to derive further signal values at an integer multiple sampling rate thereof, the integer multiple sampling rate falling within a range that is a greater than 0.5 and less than two times an integer multiple of the second sampling rate; employing a method of weighted average interpolation or decimation to derive from the signal values occurring at an integer multiple sampling rate of the first sampling rate, signal value magnitudes occurring at the integer multiple of the second signal sampling rate; and decimating the signal values at the integer multiple of the second sampling rate by an integer multiple factor to derive second sampling rate.

Abstract: The method concerns digital coding of a speech signal. The method is based on the use of a model of speech production comprising an excitation and shaping of the excitation in a filtering operation in such a manner that the order of the filtering which models the shaping of the excitation signal occurring in the vocal tract is adapted according to the speech signal to be coded. By means of the method it is possible to achieve a total modelling for the speech signal--and thus efficient speech coding--which is better than methods using fixed-order, model-based filtering of the speech tract. From the standpoint of the efficiency of the coding, by decreasing a needlessly large order of the filtering method, the bit rate to be used for coding the excitation signal can be increased or the bit rate resources thus freed up can be allocated for use in the error correction coding.