Abstract: A method for transferring data over a communication link from a first unit to a second unit, the link comprising a first portion across which the data is carried by the transmission of data packets at regular intervals, and a second portion between the first portion and the first unit over which the data is carried in a form that is not synchronised with the transmission of data packets over the first portion; the method comprising transmitting to the first unit synchronisation information regarding the times at which packets are to be transmitted over the first portion.

Abstract: The invention relates to a method for managing a packet switched, centralized conference call between a plurality of terminals 13. In order to enable an enhancement of the user comfort, it is proposed that the method comprises at a conference call server 12 receiving data packets from all terminals 13. Based on these data packets, then at least one terminal 13 currently providing voice data is determined. In a next step, the data received in the data packets is mixed, and the mixed data is inserted into new data packets together with at least one identifier associated to one of the terminals 13 which were determined to provide voice data, such that the at least one identifier can be distinguished from any other information in the data packets. Finally, the new data packets are transmitted to terminals 13 participating in the conference call.

Abstract: A system and method for providing improved adaptive multi-rate wideband (AMR-WB) discontinuous transmission (DTX) synchronization. According to various embodiments, an indication on the start of the inactive speech period is signalled to the decoder via a voice activity detection (VAD) flag a predetermined number of frames before the DTX period will start, i.e., before the SID_FIRST frame is received. When the VAD flag indicates active speech, or when the VAD flag has been set to zero less than the predetermined number of frames ago, the received NO_DATA frame can be classified with a high degree of reliability as active speech, i.e., considered as transmitter, network or terminal-initiated signalling, and can be substituted by a SPEECH_LOST frame. When the VAD flag was set to zero eight frames ago or earlier, the NO_DATA frame is classified as DTX.

Abstract: The invention relates to a method for managing a packet switched, centralized conference call between a plurality of terminals 13. In order to enable an enhancement of the user comfort, it is proposed that the method comprises at a conference call server 12 receiving data packets from all terminals 13. Based on these data packets, then at least one terminal 13 currently providing voice data is determined. In a next step, the data received in the data packets is mixed, and the mixed data is inserted into new data packets together with at least one identifier associated to one of the terminals 13 which were determined to provide voice data, such that the at least one identifier can be distinguished from any other information in the data packets. Finally, the new data packets are transmitted to terminals 13 participating in the conference call. The invention relates equally to a corresponding server and to a corresponding terminal.

Abstract: A device is disclosed that makes packetized and encoded speech data audible to a listener, as is a method for operating the device. The device includes a unit for generating a synchronization request for reducing an amount of synchronization delay, and further includes a speech decoder that is responsive to the synchronization delay adjustment request for executing a time-warping operation for one of lengthening or shortening a duration of a speech frame. In one embodiment the speech decoder comprises a code excited linear prediction (CELP) speech decoder, and the CELP decoder time-warping operation is applied to a reconstructed excitation signal u(k) to derive a time-warped reconstructed signal uw(k).

Abstract: This invention relates to methods, a computer program product and apparatuses in the context of frame buffering. A buffering time for one or more frames received by a frame buffer is determined based on a specific buffering time associated with a specific frame and on information representative of a specific amount of data stored in the frame buffer.

Abstract: A device is disclosed that makes packetized and encoded speech data audible to a listener, as is a method for operating the device. The device includes a unit for generating a synchronization request for reducing an amount of synchronization delay, and further includes a speech decoder that is responsive to the synchronization delay adjustment request for executing a time-warping operation for one of lengthening or shortening a duration of a speech frame. In one embodiment the speech decoder comprises a code excited linear prediction (CELP) speech decoder, and the CELP decoder time-warping operation is applied to a reconstructed excitation signal u(k) to derive a time-warped reconstructed signal uw(k). The time-warped reconstructed signal uw(k) is input to a Linear Predictor (LP) synthesis filter to derive a CELP decoder time-warped output signal y^w(k).

Abstract: A method and related apparatus comprising: buffering an encoded audio input signal comprising at least one combined signal of a plurality of audio channels and one or more corresponding sets of side information parameters describing a multi-channel sound image; changing the length of at least one audio frame of said combined signal by adding or removing a segment of said combined signal; modifying said one or more sets of side information parameters with a change corresponding to the change in the length of said at least one audio frame of said combined signal; and transferring said at least one audio frame of said combined signal with a changed length and said modified one or more sets of side information parameters to a further processing unit.

Abstract: This invention relates to a method, a computer program product, apparatuses and a system for controlling a length of a frame buffer. The frame buffer is comprised in a receiver and buffers frames that are transmitted by a transmitter according to a frame transmission scheme and received at the receiver. The length of the frame buffer is controlled under consideration of a change in the frame transmission scheme.

Abstract: This invention relates to a method, a computer program product, apparatuses and a system for controlling a length of a frame buffer. The frame buffer is comprised in a receiver and buffers frames that are transmitted by a transmitter according to a frame transmission scheme and received at the receiver. The length of the frame buffer is controlled under consideration of a change in the frame transmission scheme.

Abstract: The present invention provides methods and apparatus for adjusting an algorithmic time delay of a signal encoder. An input signal is sampled at a predetermined sampling rate. When look-ahead operation is initiated, the algorithmic time delay is increased by the look-ahead time duration. When look-ahead operation is terminated, the algorithmic time delay is decreased by the look-ahead time duration. A set of input signal samples is aligned in accordance with the algorithmic time delay, and an output signal that is representative of the set of signal samples is formed. A first signal segment is added to an input signal waveform when the look-ahead operation is initiated, and a second signal segment is removed from the input signal waveform when the look-ahead operation is terminated. Pointers that point to a beginning of the current frame and to new input signal samples are adjusted when the operational mode changes.

Abstract: For supporting a decoding of encoded frames, which belong to a sequence of frames received via a packet switched network, it is detected whether a particular encoded frame has been received after a scheduled decoding time for the particular encoded frame and before a scheduled decoding time for a next encoded frame. In case the particular encoded frame is detected to have been received after its scheduled decoding time and before the scheduled decoding time for the next encoded frame, the particular encoded frame is re-scheduled to be decoded at the scheduled decoding time for the next encoded frame.

Abstract: A system and method for efficiently implementing redundancy management in speech coding applications. According to various embodiments of the present invention, a sending device selects a redundant transmission level that is suitable for a current transmission channel condition while, at the same time, selecting the most suitable bit rate from the available codec mode set. With embodiments of the present invention, the redundant transmission level that is used is always optimal with regard to the selected codec modes, and no re-negotiation of the codec is needed.

Abstract: For enhancing the performance of an adaptive jitter buffer, a desired amount of adjustment of a jitter buffer is determined at a first device using as a parameter an estimated delay. The delay comprises at least an end-to-end delay in at least one direction in a conversation. For this conversation, speech signals are transmitted in packets between the first device and a second device via a packet switched network. An adjustment of the jitter buffer is then performed based on the determined amount of adjustment.

Abstract: Packets for a discontinuous transmission of a speech signal via a packet switched network may be provided in shorter transmission intervals during an active state and in longer transmission intervals during an inactive state. The active state may be selected whenever a speech signal comprises a speech burst, optionally with a hangover period after a respective speech burst. For enhancing the control of an adaptive jitter buffer at a receiver at the beginning of a respective transmission session, an active state is enforced in addition for a predetermined period at a beginning of a transmission session, irrespective of a presence of speech bursts. In case hangover periods are used, the length of the predetermined period exceeds the length of these hangover periods.

Abstract: A method and device perform error detection with partial checksum coverage by using a first transport protocol, wherein interface means are provided for requesting from a second lower-level transport protocol at least one lower-level header field to be subjected to checksum calculation over a predetermined portion of a data packet of the first transport protocol and the requested lower-level header fields. A checksum-based error processing function of the lower-level second transport protocol is disabled during the transmission of the data packet. Thereby, a checksum with partial coverage can be carried inside a higher-level data packet to provide Unequal Error Detection (UED) for error tolerant applications without requiring modifications of lower layer protocols.

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: In order to improve the audio quality of an audio signal including comfort noise, a time scaling is performed as an integral part of a comfort noise signal synthesis.

Abstract: Encoding audio signals with selecting an encoding mode for encoding the signal categorizing the signal into active segments having voice activity and non-active segments having substantially no voice activity by using categorization parameters depending on the selected encoding mode and encoding at least the active segments using the selected encoding mode.

Abstract: A method, a chipset, a receiver, a transmitter, an electronic device and a system for enabling a control of jitter management of an audio signal is described, wherein the audio signal is distributed to a sequence of frames that are received via a packet switched network, the received frames comprising active audio frames and non-active audio frames, wherein a concatenation of subsequent active audio frames represents an active audio burst, wherein a discrete information of audio activity of the audio signal via the packet switched network is received, the end of an active audio burst is determined based on the received discrete information of audio activity, and wherein jitter compensation of the received frames is controlled on the basis of the determined end of an active audio burst. The invention further relates to a corresponding software program product storing a software code for controlling jitter management of an audio signal.